- sr-users - kamailio.org

Kamailio start failed
by Roman Dmytriv 27 Jun '11

27 Jun '11

Hello friends! Having trouble to start newly installed Kamailio This is what I got: root@debian:/home/user# /etc/init.d/kamailio status Status of kamailio: kamailio is running. root@debian:/home/user# kamctl start INFO: Starting Kamailio : ERROR: PID file /var/run/kamailio.pid does not exist -- Kamailio start failed root@debian:/home/user# Any ideas will appreciated Regards Roman

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changing script (perl, lua ...) without kamailio restart
by Klaus Darilion 27 Jun '11

27 Jun '11

Hi! Does any of the script interpreters (perl, lua, python ....) allow to use a modified shell script without restarting Kamailio? Thanks Klaus

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The 15000th development commit
by Daniel-Constantin Mierla 27 Jun '11

27 Jun '11

Hello, I just wanted to mark the 15000th development commit in our project (counted only on GIT master branch). It happened several days ago and it was done by the same person that did the first commit, Andrei: git log --pretty=format:"%h%x09%an%x09%ad%x09%s" --reverse | head -1 512dcd9 Andrei Pelinescu-Onciul Mon Sep 3 21:27:11 2001 +0000 Initial revision The 15000th one is: git log --pretty=format:"%h%x09%an%x09%ad%x09%s" --reverse | head -15000 | tail -1 8a90dd3 Andrei Pelinescu-Onciul Sat Jun 11 11:24:05 2011 +0200 core: remove unused variables + coding style I published the news about it on project sites as well: * http://asipto.com/u/37 * http://asipto.com/u/38 Now it's the time to look forward to the 20000th commit! Enjoy the summer, autumn comes with v3.2.0, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

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an idea - the SIP browser - elsipo
by Daniel-Constantin Mierla 24 Jun '11

24 Jun '11

Hello, I want to present an idea and see if it is something people find benefits with such approach. Some time ago, due to existence of only SIP client applications (aka softphones) that have static GUI, which were designed for generic purposes, do not present what servces the currently used provider offers nor are easy to customize, I played with the idea of a SIP application with the GUI in HTML/CSS/Javascript. Being busy with Kamailio, the whole thing was postponed, but during last days I took the time and updated the code to latest developing tools (QT/Webkit and pjsip) and made it public on github, see: * http://asipto.com/u/3c A screenshot and some details in the wiki at: * http://asipto.com/u/3b The UI can be pulled from a remote site via http/https for example. The approach was to have a framework to render the UI, used QT/Webkit for that (cross platform, Linux, Mac OS X, Windows) and exported SIP API to Javascript. I worked with pjsip, but the system is plugin based, so pjsip plugin can be replaced, plugins for other things (not only sip/voip) can be written easily. The code is just for demo/prototyping, with the goal of showing feasibility. At this moment can be used to make/receive calls and send/receive instant messaging. In addition, content of UI can be carried by SIP messages - the examples added so far is to request some server statistics when clicking on a link of UI and the SIP server pushing periodically some info to a section of UI. Personally I don't have any plan to go into client side development, still the idea was too appealing to try and I thought would give lot of value and enable new features on SIP services to make user experience more pleasant. Therefore the target is to see if any of you think is something that worth pursuing and if yes, get people on board to develop it further from prototype phase. In the wiki I listed a part of my wish list with current VoIP/SIP services that would be trivial to implement with this approach. The naming as SIP browser instead of softphone hopefully suggest also the purpose of the application - to be able to browse the services offered by provider at that time and the provider to be able to push content back to application (on demand or periodically). Development was done on Mac OS X, a tarball with binaries for this OS is available for download on github - compiling for other OSes should go pretty straightforward. At the moment the idea poped up there was no such approach that I could find (at least not in open source space), did I miss something, are you aware of any similar approach? What is your opinion overall? Thanks, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

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Dispatcher module for INVITE
by Giuseppe Carella 24 Jun '11

24 Jun '11

Hi all, I'm using kamailio dispatcher module between the OpenIMSCore and sailfin. Normally, without the dispatcher, all works fine. From the IMS Bench I can see that there are many successful call, but when I use the dispatcher module I'm having a lot of retransmission in the first part of the INVITE protocol. I'm using the default configuration file: # ----------------- setting module-specific parameters --------------- # -- dispatcher params -- modparam("dispatcher", "list_file", "../etc/dispatcher.list") # modparam("dispatcher", "force_dst", 1) route{ if ( !mf_process_maxfwd_header("10") ) { sl_send_reply("483","To Many Hops"); drop(); }; ds_select_dst("1", "0"); forward(); # t_relay(); } I tried also the t_relay() function instead of the forward command, but the problem is the same. Thanks in advance. Bests, Giuseppe

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Don't publish Is not supposed the e-mail should be Hiden
by Omar 24 Jun '11

24 Jun '11

I really don't want to share my e-mail publicly, i remember there was an option before. O

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Remote Party ID Header
by dotnetdub 23 Jun '11

23 Jun '11

Hi All, If an endpoint is sending us the remote party ID header, I know how to extract $(re{uri.user}) - how would I check if the Privacy Flag was true in Kamailio script? Version 1.41 Thanks, Brian

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Re: [SR-Users] Kamailio doesn't hang up upon IP connectity loss to SIP endpoint
by Brett Woollum 23 Jun '11

23 Jun '11

Hi Carsten, Thanks for the tip. All audio is going through RTPProxy on the Kamailio server, not directly to Asterisk. I will look into that patch. Thanks! Brett ----- Original Message ----- From: "Carsten Bock" <carsten(a)ng-voice.com> To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org> Sent: Thursday, June 23, 2011 12:46:11 AM GMT -08:00 US/Canada Pacific Subject: Re: [SR-Users] Kamailio doesn't hang up upon IP connectity loss to SIP endpoint Hi, another solution might be, to either configure an RTP-Timeout on the Asterisk (if you send your calls through the asterisk anyway). You might also consider using the RTPProxy with the patch in the sip-router-repository. With the patch, the RTPProxy will trigger a teardown of calls (via XML-RPC) if the RTP-Session has a timeout. Carsten 2011/6/23 Brett Woollum <brett(a)woollum.com>: > Hi Alex, > > Thanks for this information. I've started researching the session-timer > capabilities in Asterisk, and I think that's my solution. I've already > implemented it on a test system and it works well, except that it's using > reINVITES to update as opposed to UPDATE messages, resulting in chops in the > audio every so often. I'll research this further though. > > Thanks again! > Brett > > ----- Original Message ----- > From: "Alex Balashov" <abalashov(a)evaristesys.com> > To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) -Users > Mailing List" <sr-users(a)lists.sip-router.org> > Cc: sr-users(a)lists.sip-router.org > Sent: Wednesday, June 22, 2011 10:22:18 PM GMT -08:00 US/Canada Pacific > Subject: Re: [SR-Users] Kamailio doesn't hang up upon IP connectity loss to > SIP endpoint > > This is a complex topic. There is no way for a proxy like Kamailio to > detect this scenario per se. Kamailio reacts to and forwards signaling > events. If an endpoint disappears, it won't send any of those to indicate > that it has gone away. How would Kamailio know? Media stream timeout? > Kamailio doesn't relay media. > Your Kamailio-side solution is a dialog timeout, requiring use of > dialog-stateful tracking using the dialog module. But that will time out > calls indiscriminately, so you need to make it long enough to not anger your > users but short enough to be useful. > Your endpoint solution is SIP Session Timers. > > -- > 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/ > On Jun 23, 2011, at 1:10 AM, Brett Woollum <brett(a)woollum.com> wrote: > > Hello, > > We are running Kamailio as a registration point for our SIP phones, which > then interacts with Asterisk. SIP registrations are processed by Kamailio, > but everything else is passed to Asterisk. The Kamailio configuration is > close to the article at: > http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb. > Everything seems to be working well, until today. > > I found several calls today that were still connected to our provider, even > though our SIP phones were not active. There were three calls with timers at > 9 hours and counting. We had some IP connectivity issues earlier today, and > I'm wonder if it's related. > > If a SIP phone was connected and on a call (through kamailio), and the > kamailio/asterisk servers became unreachable, the SIP phones will drop the > call. But, it appears that kamailio/asterisk never drop the call in this > case, and the call stays live with the carrier. I had to manually kill the > calls by command prompt. > > What's the best way to handle this? Is there a way to have kamailio or > asterisk poll the phone to see if it's still on the call or something? How > can I give visibility to asterisk or kamailio so the calls are always > dropped properly? I don't want to run up a large bill because of calls that > didn't terminate when they should have. > > Thanks! > Brett > > _______________________________________________ > 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 > _______________________________________________ > 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 http://www.ng-voice.com mailto:carsten@ng-voice.com Schomburgstr. 80 22767 Hamburg Germany Mobile +49 179 2021244 Office +49 40 34927219 Fax +49 40 34927220 _______________________________________________ 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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Re: [SR-Users] Kamailio doesn't hang up upon IP connectity loss to SIP endpoint
by Brett Woollum 23 Jun '11

23 Jun '11

Hi Alex, Thanks for this information. I've started researching the session-timer capabilities in Asterisk, and I think that's my solution. I've already implemented it on a test system and it works well, except that it's using reINVITES to update as opposed to UPDATE messages, resulting in chops in the audio every so often. I'll research this further though. Thanks again! Brett ----- Original Message ----- From: "Alex Balashov" <abalashov(a)evaristesys.com> To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) -Users Mailing List" <sr-users(a)lists.sip-router.org> Cc: sr-users(a)lists.sip-router.org Sent: Wednesday, June 22, 2011 10:22:18 PM GMT -08:00 US/Canada Pacific Subject: Re: [SR-Users] Kamailio doesn't hang up upon IP connectity loss to SIP endpoint This is a complex topic. There is no way for a proxy like Kamailio to detect this scenario per se. Kamailio reacts to and forwards signaling events. If an endpoint disappears, it won't send any of those to indicate that it has gone away. How would Kamailio know? Media stream timeout? Kamailio doesn't relay media. Your Kamailio-side solution is a dialog timeout, requiring use of dialog-stateful tracking using the dialog module. But that will time out calls indiscriminately, so you need to make it long enough to not anger your users but short enough to be useful. Your endpoint solution is SIP Session Timers. -- 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/ On Jun 23, 2011, at 1:10 AM, Brett Woollum < brett(a)woollum.com > wrote: Hello, We are running Kamailio as a registration point for our SIP phones, which then interacts with Asterisk. SIP registrations are processed by Kamailio, but everything else is passed to Asterisk. The Kamailio configuration is close to the article at: http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb . Everything seems to be working well, until today. I found several calls today that were still connected to our provider, even though our SIP phones were not active. There were three calls with timers at 9 hours and counting. We had some IP connectivity issues earlier today, and I'm wonder if it's related. If a SIP phone was connected and on a call (through kamailio), and the kamailio/asterisk servers became unreachable, the SIP phones will drop the call. But, it appears that kamailio/asterisk never drop the call in this case, and the call stays live with the carrier. I had to manually kill the calls by command prompt. What's the best way to handle this? Is there a way to have kamailio or asterisk poll the phone to see if it's still on the call or something? How can I give visibility to asterisk or kamailio so the calls are always dropped properly? I don't want to run up a large bill because of calls that didn't terminate when they should have. Thanks! Brett _______________________________________________ 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

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Kamailio doesn't hang up upon IP connectity loss to SIP endpoint
by Brett Woollum 23 Jun '11

23 Jun '11

Hello, We are running Kamailio as a registration point for our SIP phones, which then interacts with Asterisk. SIP registrations are processed by Kamailio, but everything else is passed to Asterisk. The Kamailio configuration is close to the article at: http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb. Everything seems to be working well, until today. I found several calls today that were still connected to our provider, even though our SIP phones were not active. There were three calls with timers at 9 hours and counting. We had some IP connectivity issues earlier today, and I'm wonder if it's related. If a SIP phone was connected and on a call (through kamailio), and the kamailio/asterisk servers became unreachable, the SIP phones will drop the call. But, it appears that kamailio/asterisk never drop the call in this case, and the call stays live with the carrier. I had to manually kill the calls by command prompt. What's the best way to handle this? Is there a way to have kamailio or asterisk poll the phone to see if it's still on the call or something? How can I give visibility to asterisk or kamailio so the calls are always dropped properly? I don't want to run up a large bill because of calls that didn't terminate when they should have. Thanks! Brett

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apt-get install doesn't work
by Roman Dmytriv 23 Jun '11

23 Jun '11

Hello, I was trying to install kamailio using your instruction from http://kb.asipto.com/kamailio:skype-like-service-in-less-than-one-hour I run Debian Squeeze. Imported apt repository keys, created this file </etc/apt/sources.list.d/kamailio.list> with this information in it <deb http://deb.kamailio.org/kamailio31-nightly squeeze main> Now when I run apt-get install kamailio kamailio-mysql-modules kamailio-tls-modules I got this response: root@s2:/etc/apt/sources.list.d# apt-get install kamailio kamailio-mysql-modules kamailio-tls-modules Reading package lists... Done Building dependency tree Reading state information... Done E: Unable to locate package kamailio E: Unable to locate package kamailio-mysql-modules E: Unable to locate package kamailio-tls-modules Can you give me a hint what could be wrong? Regards

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OT: are there still asymmetric SIP clients?
by Klaus Darilion 22 Jun '11

22 Jun '11

Hi! Just a small survey: Are you still using NAT detection checks? I prefer to apply NAT traversal always (never trust the contact header) which of course only works if all the clients are symmetric. Thus - do you still see asymmetric SIP clients? Thanks Klaus

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Kamailio/Asterisk/FreeSWitch dinner in Madrid - Thursday next week
by Olle E Johansson 22 Jun '11

22 Jun '11

Next week I'll be in the hot city of Madrid doing Asterisk/Kamailio training - The SIP master class. Maybe we can organize a voip nerd dinner on Thursday evening? If you're interested, please e-mail me off list and I'll send out more details later. Greetings /Olle PS. E-mail off list :-)

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Ukulele Kamailio Jingle
by Daniel-Constantin Mierla 22 Jun '11

22 Jun '11

Hello, a good friend, Javi Beltran, recorded a composition playing Ukulele (aka, ukelele - Hawaiian instrument - http://en.wikipedia.org/wiki/Ukulele). The jingle is available at: * http://asipto.com/u/39 If you like it, feel free to use it in relation with Kamailio project. Javi's compositions are open sourced, available at: * http://www.javibeltran.com/?lang=en I posted a news about it on project's site: * http://www.kamailio.org/w/2011/06/ukulele-kamailio-jingle/ If you or your friends have done any artistic work (music, graphics, a.s.o.) related to Kamailio, feel free to send it over and I will make them available on project's site. Thanks, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

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audio issue in same nat.
by †ஐû●чƃцίш† Goh 22 Jun '11

22 Jun '11

Hi List, below is my setup.. rtpproxy and kamailio in one PC with 2 nic. (ppp0 with public IP[60.49.119.XX] and eth1 with private IP[192.168.2.3])and asterisk is on another PC with private IP[192.168.2.23] i use realtime integration for kamailio and asterisk.http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6… i have two yealink hardphone ext 101(ip 192.168.1.200) and 102(ip 192.168.1.132) and a softphone ext 103 registered successful.both hardphone are behind same nat (175.136.221.XX)and softphone ext 103(ip 10.129.138.225) behind nat also (113.210.97.XX) ul show:database engine 'MYSQL' loadedControl engine 'FIFO' loadedentering fifo_cmd ul_dumpDomain:: location table=512 records=3 max_slot=1 AOR:: 102 Contact:: sip:102@175.136.221.XX:5062 Q= Expires:: 3109 Callid:: 2043273564(a)175.136.221.241 Cseq:: 4 User-agent:: T22 7.3.0.50 Received:: sip:175.136.221.241:1039 State:: CS_SYNC Flags:: 0 Cflag:: 192 Socket:: udp:60.49.119.69:5060 Methods:: 16383 AOR:: 103 Contact:: sip:103@113.210.97.XX:58776;transport=UDP;ob Q= Expires:: 294 Callid:: oa8Pqx3mR.SVnzAVEYHTwVKZE8CbpY9l Cseq:: 27626 User-agent:: v1.0.0/iPhone State:: CS_NEW Flags:: 0 Cflag:: 0 Socket:: udp:60.49.119.69:5060 Methods:: 8143 AOR:: 101 Contact:: sip:101@175.136.221.XX:5062 Q= Expires:: 1738 Callid:: 451417581(a)175.136.221.241 Cseq:: 2 User-agent:: T20 9.41.0.80 State:: CS_SYNC Flags:: 0 Cflag:: 0 Socket:: udp:60.49.119.69:5060 Methods:: 16383FIFO command was::ul_dump:openser_receiver_17783 103 try to call 102 and 101 work fine. 101 and 102 try call 103 also fine.when 101 call 102 it work fine but when 102 call 101 there is no audio for both side.102 call 101 wireshark capture on 102 sidekeep send rtp but no receive.192.168.1.132 -> 60.49.119.XX RTP when capture on 101 side.keep send rtp but no receive.192.168.1.200 -> 60.49.119.XX RTP and also when 101 try to call into voicemail there is no audioit keep send rtp packet but to192.168.1.200 -> 192.168.2.23 RTP in kamailio.cfg#!WITH_NATlisten=60.49.119.XXlisten=192.168.2.3 # uncomment next line to do SIP NAT pinging setbflag(FLB_NATSIPPING); nat_uac_test("19")rtpproxy -l 60.49.119.XX -s udp:127.0.0.1 is running anyone can help me? how can i fix this?thanks in adv. Regards, minghon

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Re: [SR-Users] [sr-dev] kamailio 3.1.3 fail to start as a service when load the ldap.so module
by Daniel-Constantin Mierla 22 Jun '11

22 Jun '11

Thanks for reporting back, taking the chance to forward this email from sr-dev to sr-users since the thread was cross-posted to some point, for the benefit of future reads of the archive to show the solution of this case. Cheers, Daniel On 6/21/11 7:19 PM, laura testi wrote: > Found the problem after some investigation based on your suggestions. > In the openldap configuration file /etc/openldap/ldap.conf, the entry > TLS_CACERT is pointed to a certificate file in a directory which is > accessible only by root user. After I move the file to a directory > which all user can access it, now it works. I can start the kamailio > as a service with "service kamailio start" with the running user and > group is kamailio. > > Thank you all for your helps and suggestions! > > Best Regards, > Laura > > On Tue, Jun 21, 2011 at 5:44 PM, laura testi <lau.testi(a)gmail.com > <mailto:lau.testi@gmail.com>> wrote: > > The SELinux is disabled. > > > > > On Tue, Jun 21, 2011 at 5:26 PM, Andrew Pogrebennyk > <andrew.pogrebennyk(a)portaone.com > <mailto:andrew.pogrebennyk@portaone.com>> wrote: > > On 21.06.2011 17 <tel:21.06.2011%2017>:30, laura testi wrote: > > Should I need to set a some special environment variable > to load ldap.so > in the /etc/init.d/kamailio? > > > The env output for root does not confirm my theory :) I'm not > sure what is preventing kamailio from communication with LDAP, > could it be that SELinux is enabled and denying it in some way? > > -- > Sincerely, > Andrew Pogrebennyk > > > > > > _______________________________________________ > sr-dev mailing list > sr-dev(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

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kamailio 3.1.3 fail to start as a service when load the ldap.so module
by laura testi 21 Jun '11

21 Jun '11

Hi, we try to use LDAP module in kamailio, first we try only load the ldap.so only to see if it can be loaded. When we start from the command line with either "kamctl start" or directly "kamilio", the kamailio is started normally with different children. When I try to start as service with "service kamailio start" or "/etc/init.d/kamailio start". It fails with the following error in the /var/log/messages: ---------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------- Jun 21 10:28:39 kamailio3p1 kamailio: INFO: <core> [tcp_main.c:4716]: init_tcp: using epoll_lt as the io watch method (auto detected) Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: usrloc [hslot.c:53]: locks array size 512 Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: auth [auth_mod.c:312]: auth: qop set, but nonce-count (nc_enabled) support disabled Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: permissions [parse_config.c:251]: file not found: //etc/kamailio/permissions.allow Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: permissions [permissions.c:606]: default allow file (//etc/kamailio/permissions.allow) not found => empty rule set Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: permissions [parse_config.c:251]: file not found: //etc/kamailio/permissions.deny Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: permissions [permissions.c:615]: default deny file (//etc/kamailio/permissions.deny) not found => empty rule set Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: ldap [ldap_mod.c:246]: OpenLDAP - 20343 Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_init.c:511]: tls: _init_tls_h: compiled with openssl version "OpenSSL 0.9.8e-fips-rhel5 01 Jul 2008" (0x0090802f), kerberos support: on, compression: on Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_init.c:519]: tls: init_tls_h: installed openssl library version "OpenSSL 0.9.8e-fips-rhel5 01 Jul 2008" (0x0090802f), kerberos support: on, zlib compression: on compiler: gcc -fPIC -DOPENSSL_PIC -DZLIB -DOPENSSL_THREADS -D_REENTRANT -DDSO_DLFCN -DHAVE_DLFCN_H -DKRB5_MIT -I/usr/kerberos/include -DL_ENDIAN -DTERMIO -Wall -DMD32_REG_T=int -O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer-size=4 -m64 -mtune=generic -Wa,--noexecstack -DOPENSSL_USE_NEW_FUNCTIONS -fno-strict-aliasing -DOPENSSL_BN_ASM_MONT -DSHA1_ASM -DSHA256_ASM -DSHA512_ASM -DMD5_ASM -DAES_ASM Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_init.c:373]: tls: init_tls: disabling compression... Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: WARNING: tls [tls_init.c:564]: tls: init_tls_h: openssl kerberos malloc bug detected, kerberos support will be disabled... Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: WARNING: tls [tls_init.c:587]: tls: openssl bug #1491 (crash/mem leaks on low memory) workaround enabled (on low memory tls operations will fail preemptively) with free memory thresholds 9961472 and 4980736 bytes Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: <core> [cfg/cfg_ctx.c:411]: INFO: cfg_set_now(): tls.low_mem_threshold1 has been changed to 9961472 Jun 21 10:28:39 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: <core> [cfg/cfg_ctx.c:411]: INFO: cfg_set_now(): tls.low_mem_threshold2 has been changed to 4980736 Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: <core> [udp_server.c:178]: INFO: udp_init: SO_RCVBUF is initially 129024 Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: <core> [udp_server.c:229]: INFO: udp_init: SO_RCVBUF is finally 262142 Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: <core> [udp_server.c:178]: INFO: udp_init: SO_RCVBUF is initially 129024 Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: <core> [udp_server.c:229]: INFO: udp_init: SO_RCVBUF is finally 262142 Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:176]: TLSs<default>: tls_method=9 Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:188]: TLSs<default>: certificate='//etc/kamailio/FBCIDS01V01.pem' Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:195]: TLSs<default>: ca_list='//etc/kamailio/tiglobalca.pem' Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:202]: TLSs<default>: crl='(null)' Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:206]: TLSs<default>: require_certificate=0 Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:213]: TLSs<default>: cipher_list='(null)' Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:220]: TLSs<default>: private_key='//etc/kamailio/FBCIDS01V01KEY.pem' Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:224]: TLSs<default>: verify_certificate=0 Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:227]: TLSs<default>: verify_depth=9 Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:544]: TLSs<default>: No client certificate required and no checks performed Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:597]: tls: set_ssl_options: detected openssl version (90802f) has the SSL_OP_TLS_BLOCK_PADDING bug, but compression is disabled so no workaround is needed Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:176]: TLSc<default>: tls_method=9 Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:188]: TLSc<default>: certificate='(null)' Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:195]: TLSc<default>: ca_list='(null)' Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:202]: TLSc<default>: crl='(null)' Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:206]: TLSc<default>: require_certificate=0 Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:213]: TLSc<default>: cipher_list='(null)' Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:220]: TLSc<default>: private_key='(null)' Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:224]: TLSc<default>: verify_certificate=0 Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:227]: TLSc<default>: verify_depth=9 Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:547]: TLSc<default>: Server MAY present invalid certificate Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:597]: tls: set_ssl_options: detected openssl version (90802f) has the SSL_OP_TLS_BLOCK_PADDING bug, but compression is disabled so no workaround is needed Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:176]: TLSs<10.10.5.214:5061>: tls_method=12 Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:188]: TLSs<10.10.5.214:5061>: certificate='//etc/kamailio/FBCIDS01V01.pem' Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:195]: TLSs<10.10.5.214:5061>: ca_list='//etc/kamailio/tiglobalca.pem' Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:202]: TLSs<10.10.5.214:5061>: crl='(null)' Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:206]: TLSs<10.10.5.214:5061>: require_certificate=0 Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:213]: TLSs<10.10.5.214:5061>: cipher_list='(null)' Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:220]: TLSs<10.10.5.214:5061>: private_key='//etc/kamailio/FBCIDS01V01KEY.pem' Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:224]: TLSs<10.10.5.214:5061>: verify_certificate=0 Jun 21 10:28:40 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:227]: TLSs<10.10.5.214:5061>: verify_depth=9 Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:544]: TLSs<10.10.5.214:5061>: No client certificate required and no checks performed Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: tls [tls_domain.c:597]: tls: set_ssl_options: detected openssl version (90802f) has the SSL_OP_TLS_BLOCK_PADDING bug, but compression is disabled so no workaround is needed Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8786]: ERROR: ldap [ldap_connect.c:166]: [sipaccounts]: ldap bind failed: Can't contact LDAP server Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8786]: ERROR: ldap [ldap_mod.c:182]: [sipaccounts]: failed to connect to LDAP host(s) Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8786]: ERROR: <core> [sr_module.c:832]: init_mod_child(): Error while initializing module ldap (/usr/lib64/kamailio/modules_k/ldap.so) Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8786]: ERROR: <core> [pt.c:337]: ERROR: fork_process(): init_child failed for process 1, pid 8786, "udp receiver child=0 sock=127.0.0.1:5060" Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8786]: : <core> [main.c:1546]: main_loop: Cannot fork Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8788]: ERROR: ldap [ldap_connect.c:166]: [sipaccounts]: ldap bind failed: Can't contact LDAP server Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8788]: ERROR: ldap [ldap_mod.c:182]: [sipaccounts]: failed to connect to LDAP host(s) Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8788]: ERROR: <core> [sr_module.c:832]: init_mod_child(): Error while initializing module ldap (/usr/lib64/kamailio/modules_k/ldap.so) Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8788]: ERROR: <core> [pt.c:337]: ERROR: fork_process(): init_child failed for process 2, pid 8788, "udp receiver child=1 sock=127.0.0.1:5060" Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8788]: : <core> [main.c:1546]: main_loop: Cannot fork Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8774]: ALERT: <core> [main.c:738]: child process 8786 exited normally, status=255 Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8774]: ALERT: <core> [main.c:738]: child process 8788 exited normally, status=255 Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8774]: INFO: <core> [main.c:756]: INFO: terminating due to SIGCHLD Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8801]: INFO: <core> [main.c:807]: INFO: signal 15 received Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8799]: INFO: <core> [main.c:807]: INFO: signal 15 received Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8797]: INFO: <core> [main.c:807]: INFO: signal 15 received Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8795]: INFO: <core> [main.c:807]: INFO: signal 15 received Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8793]: INFO: <core> [main.c:807]: INFO: signal 15 received Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8790]: INFO: <core> [main.c:807]: INFO: signal 15 received Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8812]: INFO: <core> [main.c:807]: INFO: signal 15 received Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8808]: INFO: <core> [main.c:807]: INFO: signal 15 received Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8810]: INFO: <core> [main.c:807]: INFO: signal 15 received Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8814]: INFO: <core> [main.c:807]: INFO: signal 15 received Jun 21 10:28:41 kamailio3p1 /usr/sbin/kamailio[8816]: INFO: <core> [main.c:807]: INFO: signal 15 received ... -------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------- Without ldap.so loaded, we can start the kamailio as a service normally with "service kamailio start". The platform running kamailio is: RHEL5.6_X64 Can you help us please? Because we want to start the kamailio as a service from the boot Thanks a lot in advanced! Best Regards, Laura

4 10

an IDEA
by afshin afzali 21 Jun '11

21 Jun '11

Hi Guys, I need to integrate a SIP processing environment to H.248 and SS7/ ISUP links. For those communication links, OpenSS7 and Erlang/OTP MEGACO are good solutions. My idea is to write applications over OTP and SIP-Router to be able to bridge non-SIP protocols to SIP! On OTP the way is clear, but on SIP-Router I need to be able to generate / terminate SIP transactions. I'll appreciate every comment. BEST, -- afshin

3 3

Security in proxy
by Anto 20 Jun '11

20 Jun '11

Hi After reading the modules pike, pipelimit, etc. I wanted to know what measures can be used in the proxy, because like me, there will be more people interested ;-). We see that the module pike is a good security measure, but for users with many channels, we used the configuration of users with few channels (could we discriminate and make different rules for each?). If we have the proxy settings for normal users and have trunk configuration as well, I guess we also mark the traffic of the trunk to see it too aggressive about the user. As you might know if discriminate different types of users according to their traffic, one solution I can think of is to have different proxies for each user type (trunk, normal, etc). Is there any way to implement something that detects unusual behavior on the user (referred to countries that do not ever called, excessive traffic, etc)?. I ask not to try to implement something that exists or perhaps someone can advise me how to do so because it is very likely not get it or do it wrong :-P . Continue to seek information and if I find something to contribute, send him to the list. Can you think of some way more secure proxy (mainly user accounts)? Do some basic safety tips to take into consideration? Any guidance? My advice to avoid trying to schedule something with 100 lines of code when I can do with 5 lines? Does this indicate that I intend to do not I make it :-P ?. What I said, I will try paper on the subject and if I can find information on something (though it may seem a bad code :-( ), which publishes. Thanks ;-) Regards Anto

3 4

Keeping transactions in standby till future events
by Gustavo Garcia Bernardo 20 Jun '11

20 Jun '11

Is it possible to configure sip-router to keep a transaction for an incoming message (f.e. an INVITE) without sending a response and without forwarding it, and then reference and complete it in the future? Using pseudocode this is the logic I would like to implement: //When receiving an INVITE leave it pending if (request is INVITE) { store transactionID // This transactionID will be send somehow to the client return } //When receiving a PUBLISH referencing a pending transaction complete the forwarding if (request is PUBLISH) { transactionId = header("X-TransactionId") destination = header("X-Destination") transactions [transactionID]. forward(destination) } Thank you very much, G. Este mensaje se dirige exclusivamente a su destinatario. Puede consultar nuestra política de envío y recepción de correo electrónico en el enlace situado más abajo. This message is intended exclusively for its addressee. We only send and receive email on the basis of the terms set out at. http://www.tid.es/ES/PAGINAS/disclaimer.aspx

4 7

Help find out where to modify request uri
by Simone Felici 20 Jun '11

20 Jun '11

Hi to all, I need to edit our ser installation (OLD ser-2.0.0-rc1) implementing a check to rewrite the INVITE URI address in case a flag on a DB is set or not. I'm not proof on this and as always the solution is to find out on the production system without the possibility (for now) to upgrade to most recent version. In fact I would load the flag with the db_ops module. The scenario should be the follow: 1. a call is coming from sip or from pstn to an internal sip account 2. the ser should check if a given flag for the called account is set (db_ops) 3. if it's set it should take the destination URI (taken from location db table) and rewrite the number to another known number. i.e. from sip:123456789@IP;uniqu=... to sip:new987654321@IP;uniq=... My first issue is to identify WHERE ser execute the INVITE to the sip account and how rewrite the URI. Could someone point me to the right place and give me some suggestion? The ser.cfg used could be read from a previous post of me: http://www.mail-archive.com/sr-users@lists.sip-router.org/msg04900.html Thank's a lot! Simon

2 4

Invite replies
by Lucas Alvarez 19 Jun '11

19 Jun '11

Hi, how may I catch the reply of an invite? I did this but it seems is not working: if (method=="INVITE") { $avp(sdp_ret) = fix_nated_sdp("11"); xlog("L_ALERT","#####********###### SDP return: $avp(sdp_ret)\n"); append_hf("P-hint: sdp rewritten\r\n"); t_on_reply("REPLY_TWO"); } onreply_route("REPLY_TWO") { xlog("L_ALERT","ALERT: ################ $rU - $rm from $fu (IP:$si:$sp--------$di)\n"); return; } Any help will be appreciate it, thanks in advance. Lucas

2 1

100rel without direct reachability
by Alex Balashov 19 Jun '11

19 Jun '11

(Sorry for cross-posting from Sip-Implementors; however, upon some reflection this is really more of a Kamailio mechanics question.) Greetings, I have a scenario where endpoint A is calling endpoint B through proxy P. Endpoints A and B are on different networks that are not directly reachable--A is on the public Internet, to which proxy P is connected, but NAT'd, and B is on a private RFC1918 network on which the proxy P has an interface as well. The purpose of proxy P is to make calling A <-> B possible, while providing the appropriate far-end NAT traversal functionality for A and so on and so on. It does this by inserting the correct Record-Route headers into the INVITE from A, and otherwise by staying in the signaling path; both A and B can reach the proxy, so all requests and replies go through it. The problem I'm having is prior to full dialog establishment. Upon receipt of INVITE from A by way of proxy P, B replies with 183 + SDP and also: Require: 100rel In the 183 it provides a Contact on its private network that A obviously cannot reach: Contact: <sip:xxxxx@192.168.0.10:5060> As a result, A attempts to send PRACKs to sip:xxxxx@192.168.0.10, which are obviously not going to go anywhere. After not receiving a few of these expected PRACKs, B replies to the call with a final negative failure. The question is, is there anything I can twiddle on the proxy to make this work? The problem with PRACKs is that although they are sequential requests, they arise (in this case) prior to the point at which RR can be used to guide them through the proxy. The most obvious answer is, "Make B not Require: 100rel", of course. However, that is not an option here. 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/

3 6

HTTP POST
by Matthew Williams 16 Jun '11

16 Jun '11

Hi, I'm working on an application that needs to create an HTTP POST from within Kamailio. The utils module has the http_query function, but that is only for GET. I think I'm looking for something more like the uac module's uac_req_send -- but for http. Any suggestions? Thanks in advance!

2 1

mi_datagram usage example
by Asgaroth 16 Jun '11

16 Jun '11

Hi All, I am trying to send a mi command over the mi_datagram udp socket. I dont seem to be getting a response back from the server. I am certain that it is something I am doing wrong. I have mi_datagram loaded and listening on udp:1.1.1.1:8080. I then try to send the following dispatcher mi command over to the mi_dg interface: echo -ne ":ds_list:\n\n" | nc -w 1 -u 1.1.1.1 8080 I dont see any error in the log file, so it appears that the mi_datagram is processing the request, but I am not getting any response back. The ds_list command works properly if I run it using "kamctl fifo ds_list", or if I go via "sercmd mi_dg ds_list" that works too. Would anyone have an example bash/ksh/perl snippet that would help me on my way to communicating with this mi interface. Any thoughts comments would be greatly appreciated. Thanks

3 3

Kamailio dispatcher
by Evgeniy Spinov 16 Jun '11

16 Jun '11

Hello. I had a Kamailio version of 3.0.3 and during this time configured a failover with simple routine. In short like this: if (ds_next_domain()) { xlog(.....); if (!t_relay()) { xlog(.....); return; } return; } else { t_reply("503"); } Then I've updated to 3.1.3 and was happy enough until I've discovered that my failover is not working anymore. What is observed: - It changes $rd variable, but not changing $du variable, is it ok? - Not any of the nodes receive packet on t_relay after ds_next_domain(). - In route decision section, where t_on_reply() and t_on_failure() are located I have the same t_relay() and it works fine, cause there is no any ds_next_domains(). Is it a bug or I'm doing something wrong? Thanks in advance.

3 11

How to check a bit (1/0) of a pseudo-variable?
by Iñaki Baz Castillo 16 Jun '11

16 Jun '11

Hi, summarizing I have two pseudo-variables: $var(allowed_zones) = 5; $var(id_zone) = 2; The binary representation of 5 (3+2), assuming 8 bits lenght, is: 00000101 so bit 0 = 1, bit 1 = 0, bit 2 = 1 ... Now I just need to know if bit number 2 ($var(id_zone)) of $var(allowed_zones) is 0 or 1. Is there any function like isflagset() but for pseudo-variables? Thanks a lot. -- Iñaki Baz Castillo <ibc(a)aliax.net>

3 6

Re: [SR-Users] Use of mtree module having multiple trees in the same table
by Javier Gallart 16 Jun '11

16 Jun '11

Hello not sure what you want to accomplish, but in case it might help this is how we use the mtree module. Our 3 columns are: partition, prefix, route Partition is just a name that defines a group, this is done inside the script before calling mt_match (via a table lookup for instance). Let's say if a call comes from ip x we assign it to partition 400; if it comes from ip y we assign it to partition 401. Each partition is a different tree. Consider these entries in the tree table: 400,123456,A 401,123456,A 401,1234567,B Let's say both x and y send a call with $rU 1234567; mt_match will return A for ip x, and B for ip y; we use that information for taking further routing decisions. A common situation might be: x has a unique route for a whole country while y has that entry but has more defined routes for more specific area codes inside that country. The column route could also be for instance the setid of a group of gw's for the dispatcher module, etc... Regards Javi > > Message: 3 > > Date: Tue, 14 Jun 2011 14:27:53 +0200 > > From: I?aki Baz Castillo <ibc(a)aliax.net> > > Subject: Re: [SR-Users] Use of mtree module having multiple trees in > > the same table > > To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > > Users Mailing List" <sr-users(a)lists.sip-router.org> > > Message-ID: <BANLkTim8h563kUy0YT9pTgOSXJ6W4Qf6ww(a)mail.gmail.com> > > Content-Type: text/plain; charset=UTF-8 > > > > 2011/6/13 jose luis millan <doctore96(a)gmail.com>: > > > I wonder if anyone could help me understanding how can I use this > > > module in a scenario where multiple mtrees are defined behind the same > > > database table. > > > > > > I can't guess how to speficy to 'mt_match' the mtree I want to match > > > against, since the 'mtree' module parameter defines a specific memory > > > tree based on a specific table, but not based on a specific 'tname' > > > value. > > > > Indeed we have a table with these colums: > > > > - prefix: string to identify number destination. This is the regex > > column. > > - id_zone: integer, points to other table. This is the value column. > > - scope: integer, can be 0 or 1. > > > > So wee need two instances of mtree: > > 1) those rows with scope 0. > > 2) those rows with scope 1. > > > > Is it possible with current mtree module? or maybe we'd do better by > > creating two views of the table (one for entries with scope 0 and > > another for entries with scope 1)? > > > > Thanks a lot. > > > > -- > > I?aki Baz Castillo > > <ibc(a)aliax.net> > > > > > > > > >

2 1

Re: [SR-Users] sr-users Digest, Vol 73, Issue 40
by Javier Gallart 16 Jun '11

16 Jun '11

Hello not sure what you want to accomplish, but in case it might help this is how we use the mtree module. Our 3 columns are: partition, prefix, route Partition is just a name that defines a group, this is done inside the script before calling mt_match (via a table lookup for instance). Let's say if a call comes from ip x we assign it to partition 400; if it comes from ip y we assign it to partition 401. Each partition is a different tree. Consider these entries in the tree table: 400,123456,A 401,123456,A 401,1234567,B Let's say both x and y send a call with $rU 1234567; mt_match will return A for ip x, and B for ip y; we use that information for taking further routing decisions. A common situation might be: x has a unique route for a whole country while y has that entry but has more defined routes for more specific area codes inside that country. The column route could also be for instance the setid of a group of gw's for the dispatcher module, etc... Regards Javi Message: 3 > Date: Tue, 14 Jun 2011 14:27:53 +0200 > From: I?aki Baz Castillo <ibc(a)aliax.net> > Subject: Re: [SR-Users] Use of mtree module having multiple trees in > the same table > To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List" <sr-users(a)lists.sip-router.org> > Message-ID: <BANLkTim8h563kUy0YT9pTgOSXJ6W4Qf6ww(a)mail.gmail.com> > Content-Type: text/plain; charset=UTF-8 > > 2011/6/13 jose luis millan <doctore96(a)gmail.com>: > > I wonder if anyone could help me understanding how can I use this > > module in a scenario where multiple mtrees are defined behind the same > > database table. > > > > I can't guess how to speficy to 'mt_match' the mtree I want to match > > against, since the 'mtree' module parameter defines a specific memory > > tree based on a specific table, but not based on a specific 'tname' > > value. > > Indeed we have a table with these colums: > > - prefix: string to identify number destination. This is the regex > column. > - id_zone: integer, points to other table. This is the value column. > - scope: integer, can be 0 or 1. > > So wee need two instances of mtree: > 1) those rows with scope 0. > 2) those rows with scope 1. > > Is it possible with current mtree module? or maybe we'd do better by > creating two views of the table (one for entries with scope 0 and > another for entries with scope 1)? > > Thanks a lot. > > -- > I?aki Baz Castillo > <ibc(a)aliax.net> > > > >

1 0

t_relay_to("0x02") alternative with kamailio 3.1
by Herve Couplet 16 Jun '11

16 Jun '11

Hi, I had a setup with an old version where I used the function t_relay_to("0x02"). Following the 3.1 documentation, flag : * 0x02* - do not generate reply on internal error (NOTE: has no effect anymore) I used that function as a workaround with a pbx that stop to register when it receive an error response like 478 Unresolvable destination or 408 request timeout. I did upgrade to 3.1 to have dns srv caching. Do you know how this would be possible with kamailio 3.1. Thanks for your suggestions, hp

4 4

Operations With Variables $var- does it need to load a module
by Omar 15 Jun '11

15 Jun '11

In Kamailio 1.4.3 I have the following scenario: $var(a) = 0 $var(b) = 0 $var(a) = $(tU{s.len}); $var(b) = $var(a) - 5; xlog(" $var(b)"); the result $var(b) is always the same as $var(a) should i check if some specific module is loaded? the debugs does not show any error, and $var(a) and $var(b) are unique. Variables are working, but not the operations with the variables. regards O

3 4

200 OK Retransmission Error
by Jijo 15 Jun '11

15 Jun '11

Hi All, In an INVITE transaction, no ACK from A party and B party continue retransmission of 200 OK. I have observed that kamailio is not passing the retransmitted 200 OK message from B party to the Application(script) after 3 seconds. Instead kamailio directly sending to A party without passing to the script. I see in the logs that kamailio is cleaning up the transaction after 3 seconds..look for the string "DEBUG: t_lookup_request: no transaction found" before this log it is cleaning up the timer in the log "DEBUG: cleanup_uac_timers: RETR/FR timers reset" What does it mean by that?? How can we make the kamailio to pass the response to the application? Please let me know your comments.. Here is the flow. I cannot send the complete logs which is quite big. so i grep'd using the stirng "tcp_read_req: content-length|get_hdr_field|t_reply_matching|cleanup_uac_timers|RR11" from the complete log.. A kamailo B ----INVITE-->- --------------------INVITE 180 Ringing--<---------------180 Ringing 200 OK-----------<---------------200 OK 0 200 OK-----------<---------------200 OK 500ms 200 OK-----------<---------------200 OK 1s 200 OK-----------<---------------200 OK 2s < From here it is not passing to the application- script> 200 OK-----------<---------------200 OK 4s 200 OK-----------<---------------200 OK 4s 28 102:2011-06-13T19:47:37+01:00 [debug] sipserver: [26177] DEBUG: <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [50]; uri=[ sip:5616990102@192.168.243.151:5060] 36 95:2011-06-13T19:47:37+01:00 [debug] sipserver: [26177] DEBUG: <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE> 267 86:2011-06-13T19:47:37+01:00 [debug] sipserver: [26269] DEBUG: <core> [tcp_read.c:933]: tcp_read_req: content-length= 0 282 102:2011-06-13T19:47:37+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[ sip:5616990102@192.168.243.151:5060] 284 95:2011-06-13T19:47:37+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE> 285 89:2011-06-13T19:47:37+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch 0 286 90:2011-06-13T19:47:38+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1018]: DEBUG: t_reply_matching: reply matched (T=0xb0f62e50)! 292 96:2011-06-13T19:47:38+01:00 [info] sipserver: [26269] INFO: <script>: CI=1-8028(a)192.168.243.150 -RR11 - Entry M=INVITE RURI=<null> From= sip:5616990103@192.168.243.150 To=sip:5616990102@192.168.243.151:5060 RD= <null> IP=192.168.243.161 Code:180 Reason=Ringing CT=5616990102 <sip: 5616990102@192.168.243.161:5060;transport=tcp> 331 86:2011-06-13T19:47:39+01:00 [debug] sipserver: [26269] DEBUG: <core> [tcp_read.c:933]: tcp_read_req: content-length= 191 346 102:2011-06-13T19:47:39+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[ sip:5616990102@192.168.243.151:5060] 348 95:2011-06-13T19:47:39+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE> 349 89:2011-06-13T19:47:39+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch 0 350 90:2011-06-13T19:47:39+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1018]: DEBUG: t_reply_matching: reply matched (T=0xb0f62e50)! 356 96:2011-06-13T19:47:39+01:00 [info] sipserver: [26269] INFO: <script>: CI=1-8028(a)192.168.243.150 -RR11 - Entry M=INVITE RURI=<null> From= sip:5616990103@192.168.243.150 To=sip:5616990102@192.168.243.151:5060 RD= <null> IP=192.168.243.161 Code:200 Reason=OK CT=5616990102 <sip:5616990102 @192.168.243.161:5060;transport=tcp> 398 89:2011-06-13T19:47:39+01:00 [debug] sipserver: [26269] DEBUG: tm [t_reply.c:1467]: DEBUG: cleanup_uac_timers: RETR/FR timers reset 406 86:2011-06-13T19:47:40+01:00 [debug] sipserver: [26269] DEBUG: <core> [tcp_read.c:933]: tcp_read_req: content-length= 191 421 102:2011-06-13T19:47:40+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[ sip:5616990102@192.168.243.151:5060] 423 95:2011-06-13T19:47:40+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE> 424 89:2011-06-13T19:47:40+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch 0 425 90:2011-06-13T19:47:40+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1018]: DEBUG: t_reply_matching: reply matched (T=0xb0f62e50)! 431 96:2011-06-13T19:47:40+01:00 [info] sipserver: [26269] INFO: <script>: CI=1-8028(a)192.168.243.150 -RR11 - Entry M=INVITE RURI=<null> From= sip:5616990103@192.168.243.150 To=sip:5616990102@192.168.243.151:5060 RD= <null> IP=192.168.243.161 Code:200 Reason=OK CT=5616990102 <sip:5616990102 @192.168.243.161:5060;transport=tcp> 481 86:2011-06-13T19:47:41+01:00 [debug] sipserver: [26269] DEBUG: <core> [tcp_read.c:933]: tcp_read_req: content-length= 191 496 102:2011-06-13T19:47:41+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[ sip:5616990102@192.168.243.151:5060] 498 95:2011-06-13T19:47:41+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE> 499 89:2011-06-13T19:47:41+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch 0 500 90:2011-06-13T19:47:41+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1018]: DEBUG: t_reply_matching: reply matched (T=0xb0f62e50)! 506 96:2011-06-13T19:47:41+01:00 [info] sipserver: [26269] INFO: <script>: CI=1-8028(a)192.168.243.150 -RR11 - Entry M=INVITE RURI=<null> From= sip:5616990103@192.168.243.150 To=sip:5616990102@192.168.243.151:5060 RD= <null> IP=192.168.243.161 Code:200 Reason=OK CT=5616990102 <sip:5616990102 @192.168.243.161:5060;transport=tcp> 556 86:2011-06-13T19:47:43+01:00 [debug] sipserver: [26269] DEBUG: <core> [tcp_read.c:933]: tcp_read_req: content-length= 191 571 102:2011-06-13T19:47:43+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[ sip:5616990102@192.168.243.151:5060] 573 95:2011-06-13T19:47:43+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE> 574 89:2011-06-13T19:47:43+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch 0 575 90:2011-06-13T19:47:43+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1018]: DEBUG: t_reply_matching: reply matched (T=0xb0f62e50)! 581 96:2011-06-13T19:47:43+01:00 [info] sipserver: [26269] INFO: <script>: CI=1-8028(a)192.168.243.150 -RR11 - Entry M=INVITE RURI=<null> From= sip:5616990103@192.168.243.150 To=sip:5616990102@192.168.243.151:5060 RD= <null> IP=192.168.243.161 Code:200 Reason=OK CT=5616990102 <sip:5616990102 @192.168.243.161:5060;transport=tcp> 631 86:2011-06-13T19:47:47+01:00 [debug] sipserver: [26269] DEBUG: <core> [tcp_read.c:933]: tcp_read_req: content-length= 191 646 102:2011-06-13T19:47:47+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[ sip:5616990102@192.168.243.151:5060] 648 95:2011-06-13T19:47:47+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE> 649 89:2011-06-13T19:47:47+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch 0 650 90:2011-06-13T19:47:47+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1052]: DEBUG: t_reply_matching: no matching transaction exists 651 90:2011-06-13T19:47:47+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1055]: DEBUG: t_reply_matching: failure to match a transaction 693 86:2011-06-13T19:47:51+01:00 [debug] sipserver: [26269] DEBUG: <core> [tcp_read.c:933]: tcp_read_req: content-length= 191 708 102:2011-06-13T19:47:51+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[ sip:5616990102@192.168.243.151:5060] 710 95:2011-06-13T19:47:51+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE> 711 89:2011-06-13T19:47:51+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch 0 712 90:2011-06-13T19:47:51+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1052]: DEBUG: t_reply_matching: no matching transaction exists 713 90:2011-06-13T19:47:51+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1055]: DEBUG: t_reply_matching: failure to match a transaction 755 86:2011-06-13T19:47:55+01:00 [debug] sipserver: [26269] DEBUG: <core> [tcp_read.c:933]: tcp_read_req: content-length= 191 770 102:2011-06-13T19:47:55+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[ sip:5616990102@192.168.243.151:5060] 772 95:2011-06-13T19:47:55+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE> 773 89:2011-06-13T19:47:55+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch 0 774 90:2011-06-13T19:47:55+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1052]: DEBUG: t_reply_matching: no matching transaction exists 775 90:2011-06-13T19:47:55+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1055]: DEBUG: t_reply_matching: failure to match a transaction 817 86:2011-06-13T19:47:59+01:00 [debug] sipserver: [26269] DEBUG: <core> [tcp_read.c:933]: tcp_read_req: content-length= 191 832 102:2011-06-13T19:47:59+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[ sip:5616990102@192.168.243.151:5060] 834 95:2011-06-13T19:47:59+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE> 835 89:2011-06-13T19:47:59+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch 0 836 90:2011-06-13T19:47:59+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1052]: DEBUG: t_reply_matching: no matching transaction exists 837 90:2011-06-13T19:47:59+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1055]: DEBUG: t_reply_matching: failure to match a transaction 879 86:2011-06-13T19:48:03+01:00 [debug] sipserver: [26269] DEBUG: <core> [tcp_read.c:933]: tcp_read_req: content-length= 191 894 102:2011-06-13T19:48:03+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[ sip:5616990102@192.168.243.151:5060] 896 95:2011-06-13T19:48:03+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE> 897 89:2011-06-13T19:48:03+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch 0 898 90:2011-06-13T19:48:03+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1052]: DEBUG: t_reply_matching: no matching transaction exists 899 90:2011-06-13T19:48:03+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1055]: DEBUG: t_reply_matching: failure to match a transaction 941 86:2011-06-13T19:48:07+01:00 [debug] sipserver: [26269] DEBUG: <core> [tcp_read.c:933]: tcp_read_req: content-length= 191 956 102:2011-06-13T19:48:07+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[ sip:5616990102@192.168.243.151:5060] 958 95:2011-06-13T19:48:07+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE> 959 89:2011-06-13T19:48:07+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch 0 960 90:2011-06-13T19:48:07+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1052]: DEBUG: t_reply_matching: no matching transaction exists 961 90:2011-06-13T19:48:07+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1055]: DEBUG: t_reply_matching: failure to match a transaction 1003 86:2011-06-13T19:48:11+01:00 [debug] sipserver: [26269] DEBUG: <core> [tcp_read.c:933]: tcp_read_req: content-length= 191 1018 102:2011-06-13T19:48:11+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[ sip:5616990102@192.168.243.151:5060] 1020 95:2011-06-13T19:48:11+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE> 1021 89:2011-06-13T19:48:11+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch 0 1022 90:2011-06-13T19:48:11+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1052]: DEBUG: t_reply_matching: no matching transaction exists 1023 90:2011-06-13T19:48:11+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1055]: DEBUG: t_reply_matching: failure to match a transaction

2 2

NAPTR priorities doesn't seem to work properly
by Iñaki Baz Castillo 15 Jun '11

15 Jun '11

Hi, I'm testing Kamailio's NAPTR resolution: version: kamailio 3.2.0-dev4 (x86_64/linux) flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 4MB DNS related configuration: dns_servers_no=2 dns_srv_lb=yes dns_try_naptr=yes dns_use_search_list=no use_dns_cache=yes use_dns_failover=yes dns_tls_preference=1 dns_tcp_preference=1 dns_udp_preference=1 As the doc says it should respect the priorities of the NAPTR records, see http://kamailio.org/dokuwiki/doku.php/core-cookbook:3.1.x#dns_sctp_pref_dns…: "To use the remote site preferences set all dns_*_pref to the same positive value (e.g. dns_udp_pref=1, dns_tcp_pref=1, dns_tls_pref=1, dns_sctp_pref=1)" So my kamailio receives an INVITE for an external domain oversip.net (no transport param neither port in the RURI) and does a t_relay. Request RURI is sip:qwe@oversip.net. According to NAPTR: ~$ host -t naptr oversip.net oversip.net has NAPTR record 5 50 "S" "SIPS+D2T" "" _sips._tcp.oversip.net. oversip.net has NAPTR record 10 50 "S" "SIP+D2T" "" _sip._tcp.oversip.net. oversip.net has NAPTR record 20 50 "S" "SIP+D2U" "" _sip._udp.oversip.net. oversip.net has NAPTR record 40 50 "S" "SIP+D2S" "" _sip._sctp.oversip.net. oversip.net has NAPTR record 50 50 "S" "SIPS+D2S" "" _sips._sctp.oversip.net. So it should try TLS over TCP first, if it fails try TCP and if it fails try UDP. However it just uses UDP, why?? Even if I set a minor value to dns_tls_preference (so higher priority I expect) it still uses UDP. Am I doing something wrong? Thanks a lot. -- Iñaki Baz Castillo <ibc(a)aliax.net>

6 29

kamailio conference calls support
by Mino Haluz 15 Jun '11

15 Jun '11

Hi, does kamailio support conference calls or it has to be implemented in some B2BUA ? If it is supported, what are the requirements (modules etc.). Thanks, Mino

2 1

302 rewrite ruri based on contact
by Omar 15 Jun '11

15 Jun '11

I need to rewrite the destination after I receive 302 I have this logic Onreply Loads in avp the information needed I try sending to failure route and to a route in both cases the $ru become <null> In the route and failureroute I did try rewritehost, rewriteuri, $ru= number+ host And append the route But $ru is always null. Inside the route or failure route If there are no 302 nor the onreply involved everything works fine Any suggestion to rewrite the ruri after a 302 is welcome I am still using Kamailo1.4.3 Omar

3 6

Modifying sip packet
by Lucas Alvarez 14 Jun '11

14 Jun '11

Hi, I want to modify the following sip packet, I want to change the SDP IP address by $di, which rule could filter this packet? Something like: [ if method("OK") and $di is RFC1918 ] ??? And with which function could I change the Ip address in the SDP content? In this case I would like to change 172.16.99.2 by 190.244.125.41. Thanks in advance, Lucas T 172.16.99.2:5060 -> 190.244.125.41:9055 [AP] SIP/2.0 200 OK. Via: SIP/2.0/UDP 172.16.99.2;branch=z9hG4bK7c5.bfc43456.0;i=1;received=172.16.99.2;rport=5060. Via: SIP/2.0/TCP 192.168.15.231:1952 ;received=190.244.125.41;branch=z9hG4bK-d8754z-be53247ec649be5b-1---d8754z-;rport=9055. Record-Route: <sip:172.16.99.2;r2=on;lr=on;ftag=b11c6a4e>. Record-Route: <sip:172.16.99.2;transport=tcp;r2=on;lr=on;ftag=b11c6a4e>. From: "Lucas"<sip:1104@67.152.18.231>;tag=b11c6a4e. To: <sip:1101@67.152.18.231>;tag=as5356a251. Call-ID: NzcwZWIxNGE1MDg5ZjVhZWRhYWNlM2RiNmMwM2M4MWM.. CSeq: 2 INVITE. Server: Asterisk PBX 1.8.4.1. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Contact: <sip:1101@172.16.99.2:5080>. Remote-Party-ID: "Gus Office One" <sip:1101@67.152.18.231 >;party=called;privacy=off;screen=no. Content-Type: application/sdp. Content-Length: 305. . v=0. o=root 1124623741 1124623741 IN IP4 172.16.99.2. s=Asterisk PBX 1.8.4.1. c=IN IP4 172.16.99.2. t=0 0. m=audio 13640 RTP/AVP 0 8 18 101. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. 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.

2 1

Use of mtree module having multiple trees in the same table
by jose luis millan 14 Jun '11

14 Jun '11

hello, I wonder if anyone could help me understanding how can I use this module in a scenario where multiple mtrees are defined behind the same database table. I can't guess how to speficy to 'mt_match' the mtree I want to match against, since the 'mtree' module parameter defines a specific memory tree based on a specific table, but not based on a specific 'tname' value. Any help is apreciated, Thanks in advance.

2 2

Fwd: 200OK Retransmission Error
by Jijo 14 Jun '11

14 Jun '11

Hi All, In an INVITE transaction, no ACK from A party and B party continue retransmission of 200 OK. I have observed that kamailio is not passing the retransmitted 200 OK message from B party to the Application(script) after 3 seconds. Instead kamailio directly sending to A party without passing to the script. I see in the logs that kamailio is cleaning up the transaction after 3 seconds..look for the string "DEBUG: t_lookup_request: no transaction found" before this log it is cleaning up the timer in the log "DEBUG: cleanup_uac_timers: RETR/FR timers reset" What does it mean by that?? How can we make the kamailio to pass the response to the application? Please let me know your comments.. Here is the flow. I cannot send the complete logs which is quite big. so i grep'd using the stirng "tcp_read_req: content-length|get_hdr_field|t_reply_matching|cleanup_uac_timers|RR11" from the complete log.. A kamailo B ----INVITE-->- --------------------INVITE 180 Ringing--<---------------180 Ringing 200 OK-----------<---------------200 OK 0 200 OK-----------<---------------200 OK 500ms 200 OK-----------<---------------200 OK 1s 200 OK-----------<---------------200 OK 2s < From here it is not passing to the application- script> 200 OK-----------<---------------200 OK 4s 200 OK-----------<---------------200 OK 4s 28 102:2011-06-13T19:47:37+01:00 [debug] sipserver: [26177] DEBUG: <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [50]; uri=[ sip:5616990102@192.168.243.151:5060] 36 95:2011-06-13T19:47:37+01:00 [debug] sipserver: [26177] DEBUG: <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE> 267 86:2011-06-13T19:47:37+01:00 [debug] sipserver: [26269] DEBUG: <core> [tcp_read.c:933]: tcp_read_req: content-length= 0 282 102:2011-06-13T19:47:37+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[ sip:5616990102@192.168.243.151:5060] 284 95:2011-06-13T19:47:37+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE> 285 89:2011-06-13T19:47:37+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch 0 286 90:2011-06-13T19:47:38+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1018]: DEBUG: t_reply_matching: reply matched (T=0xb0f62e50)! 292 96:2011-06-13T19:47:38+01:00 [info] sipserver: [26269] INFO: <script>: CI=1-8028(a)192.168.243.150 -RR11 - Entry M=INVITE RURI=<null> From= sip:5616990103@192.168.243.150 To=sip:5616990102@192.168.243.151:5060 RD= <null> IP=192.168.243.161 Code:180 Reason=Ringing CT=5616990102 <sip:5616990102@192.168.243.161:5060;transport=tcp> 331 86:2011-06-13T19:47:39+01:00 [debug] sipserver: [26269] DEBUG: <core> [tcp_read.c:933]: tcp_read_req: content-length= 191 346 102:2011-06-13T19:47:39+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[ sip:5616990102@192.168.243.151:5060] 348 95:2011-06-13T19:47:39+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE> 349 89:2011-06-13T19:47:39+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch 0 350 90:2011-06-13T19:47:39+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1018]: DEBUG: t_reply_matching: reply matched (T=0xb0f62e50)! 356 96:2011-06-13T19:47:39+01:00 [info] sipserver: [26269] INFO: <script>: CI=1-8028(a)192.168.243.150 -RR11 - Entry M=INVITE RURI=<null> From= sip:5616990103@192.168.243.150 To=sip:5616990102@192.168.243.151:5060 RD= <null> IP=192.168.243.161 Code:200 Reason=OK CT=5616990102 <sip:5616990102@192.168.243.161:5060;transport=tcp> 398 89:2011-06-13T19:47:39+01:00 [debug] sipserver: [26269] DEBUG: tm [t_reply.c:1467]: DEBUG: cleanup_uac_timers: RETR/FR timers reset 406 86:2011-06-13T19:47:40+01:00 [debug] sipserver: [26269] DEBUG: <core> [tcp_read.c:933]: tcp_read_req: content-length= 191 421 102:2011-06-13T19:47:40+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[ sip:5616990102@192.168.243.151:5060] 423 95:2011-06-13T19:47:40+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE> 424 89:2011-06-13T19:47:40+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch 0 425 90:2011-06-13T19:47:40+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1018]: DEBUG: t_reply_matching: reply matched (T=0xb0f62e50)! 431 96:2011-06-13T19:47:40+01:00 [info] sipserver: [26269] INFO: <script>: CI=1-8028(a)192.168.243.150 -RR11 - Entry M=INVITE RURI=<null> From= sip:5616990103@192.168.243.150 To=sip:5616990102@192.168.243.151:5060 RD= <null> IP=192.168.243.161 Code:200 Reason=OK CT=5616990102 <sip:5616990102@192.168.243.161:5060;transport=tcp> 481 86:2011-06-13T19:47:41+01:00 [debug] sipserver: [26269] DEBUG: <core> [tcp_read.c:933]: tcp_read_req: content-length= 191 496 102:2011-06-13T19:47:41+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[ sip:5616990102@192.168.243.151:5060] 498 95:2011-06-13T19:47:41+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE> 499 89:2011-06-13T19:47:41+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch 0 500 90:2011-06-13T19:47:41+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1018]: DEBUG: t_reply_matching: reply matched (T=0xb0f62e50)! 506 96:2011-06-13T19:47:41+01:00 [info] sipserver: [26269] INFO: <script>: CI=1-8028(a)192.168.243.150 -RR11 - Entry M=INVITE RURI=<null> From= sip:5616990103@192.168.243.150 To=sip:5616990102@192.168.243.151:5060 RD= <null> IP=192.168.243.161 Code:200 Reason=OK CT=5616990102 <sip:5616990102@192.168.243.161:5060;transport=tcp> 556 86:2011-06-13T19:47:43+01:00 [debug] sipserver: [26269] DEBUG: <core> [tcp_read.c:933]: tcp_read_req: content-length= 191 571 102:2011-06-13T19:47:43+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[ sip:5616990102@192.168.243.151:5060] 573 95:2011-06-13T19:47:43+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE> 574 89:2011-06-13T19:47:43+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch 0 575 90:2011-06-13T19:47:43+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1018]: DEBUG: t_reply_matching: reply matched (T=0xb0f62e50)! 581 96:2011-06-13T19:47:43+01:00 [info] sipserver: [26269] INFO: <script>: CI=1-8028(a)192.168.243.150 -RR11 - Entry M=INVITE RURI=<null> From= sip:5616990103@192.168.243.150 To=sip:5616990102@192.168.243.151:5060 RD= <null> IP=192.168.243.161 Code:200 Reason=OK CT=5616990102 <sip:5616990102@192.168.243.161:5060;transport=tcp> 631 86:2011-06-13T19:47:47+01:00 [debug] sipserver: [26269] DEBUG: <core> [tcp_read.c:933]: tcp_read_req: content-length= 191 646 102:2011-06-13T19:47:47+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[ sip:5616990102@192.168.243.151:5060] 648 95:2011-06-13T19:47:47+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE> 649 89:2011-06-13T19:47:47+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch 0 650 90:2011-06-13T19:47:47+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1052]: DEBUG: t_reply_matching: no matching transaction exists 651 90:2011-06-13T19:47:47+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1055]: DEBUG: t_reply_matching: failure to match a transaction 693 86:2011-06-13T19:47:51+01:00 [debug] sipserver: [26269] DEBUG: <core> [tcp_read.c:933]: tcp_read_req: content-length= 191 708 102:2011-06-13T19:47:51+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[ sip:5616990102@192.168.243.151:5060] 710 95:2011-06-13T19:47:51+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE> 711 89:2011-06-13T19:47:51+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch 0 712 90:2011-06-13T19:47:51+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1052]: DEBUG: t_reply_matching: no matching transaction exists 713 90:2011-06-13T19:47:51+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1055]: DEBUG: t_reply_matching: failure to match a transaction 755 86:2011-06-13T19:47:55+01:00 [debug] sipserver: [26269] DEBUG: <core> [tcp_read.c:933]: tcp_read_req: content-length= 191 770 102:2011-06-13T19:47:55+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[ sip:5616990102@192.168.243.151:5060] 772 95:2011-06-13T19:47:55+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE> 773 89:2011-06-13T19:47:55+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch 0 774 90:2011-06-13T19:47:55+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1052]: DEBUG: t_reply_matching: no matching transaction exists 775 90:2011-06-13T19:47:55+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1055]: DEBUG: t_reply_matching: failure to match a transaction 817 86:2011-06-13T19:47:59+01:00 [debug] sipserver: [26269] DEBUG: <core> [tcp_read.c:933]: tcp_read_req: content-length= 191 832 102:2011-06-13T19:47:59+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[ sip:5616990102@192.168.243.151:5060] 834 95:2011-06-13T19:47:59+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE> 835 89:2011-06-13T19:47:59+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch 0 836 90:2011-06-13T19:47:59+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1052]: DEBUG: t_reply_matching: no matching transaction exists 837 90:2011-06-13T19:47:59+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1055]: DEBUG: t_reply_matching: failure to match a transaction 879 86:2011-06-13T19:48:03+01:00 [debug] sipserver: [26269] DEBUG: <core> [tcp_read.c:933]: tcp_read_req: content-length= 191 894 102:2011-06-13T19:48:03+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[ sip:5616990102@192.168.243.151:5060] 896 95:2011-06-13T19:48:03+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE> 897 89:2011-06-13T19:48:03+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch 0 898 90:2011-06-13T19:48:03+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1052]: DEBUG: t_reply_matching: no matching transaction exists 899 90:2011-06-13T19:48:03+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1055]: DEBUG: t_reply_matching: failure to match a transaction 941 86:2011-06-13T19:48:07+01:00 [debug] sipserver: [26269] DEBUG: <core> [tcp_read.c:933]: tcp_read_req: content-length= 191 956 102:2011-06-13T19:48:07+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[ sip:5616990102@192.168.243.151:5060] 958 95:2011-06-13T19:48:07+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE> 959 89:2011-06-13T19:48:07+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch 0 960 90:2011-06-13T19:48:07+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1052]: DEBUG: t_reply_matching: no matching transaction exists 961 90:2011-06-13T19:48:07+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1055]: DEBUG: t_reply_matching: failure to match a transaction 1003 86:2011-06-13T19:48:11+01:00 [debug] sipserver: [26269] DEBUG: <core> [tcp_read.c:933]: tcp_read_req: content-length= 191 1018 102:2011-06-13T19:48:11+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:185]: DEBUG: get_hdr_field: <To> [65]; uri=[ sip:5616990102@192.168.243.151:5060] 1020 95:2011-06-13T19:48:11+01:00 [debug] sipserver: [26269] DEBUG: <core> [parser/msg_parser.c:165]: get_hdr_field: cseq <CSeq>: <1> <INVITE> 1021 89:2011-06-13T19:48:11+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:967]: DEBUG: t_reply_matching: hash 5084 label 418362480 branch 0 1022 90:2011-06-13T19:48:11+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1052]: DEBUG: t_reply_matching: no matching transaction exists 1023 90:2011-06-13T19:48:11+01:00 [debug] sipserver: [26269] DEBUG: tm [t_lookup.c:1055]: DEBUG: t_reply_matching: failure to match a transaction

1 1

RTP proxy
by michel freiha 14 Jun '11

14 Jun '11

Dear All, I'm using the below config plan for routing my calls...The issue that forcing rtp proxy is not working well and the rtp proxy is never forced....Can you please provide me a guidance here? if($rU=~"^00.*" ) { if(!cr_route("default", "domain.com", "$rU", "$rU", "call_id")){ sl_send_reply("403", "Not allowed"); } else { setbflag(6); route(1); } exit; ## this was missing here -bogdan!!! } else { route(LOCATION); route(RELAY); } } ############# THIS IS THE GENERIC RELAY ROUTE THAT MUST BE USED ALL THE TIME FOR RELAY route[1] { # apply any NAT changes? if(isflagset(5)||isbflagset(6)) { route(4); } exit; } route[4] { if (is_method("BYE|CANCEL")) { unforce_rtp_proxy(); } else if (is_method("INVITE")){ if (nat_uac_test("8")) { * force_rtp_proxy();* } else { force_rtp_proxy(); } xlog("L_ERR","66666666666666666666666666666666666666666"); t_on_failure("1"); t_on_reply("1"); }; The call is going through Route[1] then through Route[4], but rtp proxy is never forced...Any tips please? Regards

2 1

Meaning of empty body in NOTIFY
by Eugen Dedu 13 Jun '11

13 Jun '11

Hi, ekiga.net registrar uses kamailio 1.5.3 (yes, a bit old...) and for users who are not registered an empty NOTIFY body is returned when asked by a SUBSCRIBE. What does this mean from SIP standard point of view, and from kamailio point of view (are they identical?) I see in RFC3265/3.1.6.2: .... If the resource has no meaningful state at the time that the SUBSCRIBE message is processed, this NOTIFY message MAY contain an empty or neutral body but is difficult for me to interpret what it means. Example: I ask the presence for a user xyz who registered and quit application long time ago: SUBSCRIBE sip:xyz@ekiga.net SIP/2.0 CSeq: 1 SUBSCRIBE Via: SIP/2.0/UDP 82.238.108.175:5060;branch=z9hG4bKdabe824f-1a8e-e011-9efc-0024d693d8e8;rport User-Agent: Ekiga/3.3.1 From: <sip:eugen.dedu@ekiga.net>;tag=4888824f-1a8e-e011-9efc-0024d693d8e8 Call-ID: f602824f-1a8e-e011-9efc-0024d693d8e8@snoopy Supported: eventlist To: <sip:xyz@ekiga.net> Accept: application/pidf+xml Accept: multipart/related Accept: application/rlmi+xml Contact: <sip:eugen.dedu@82.238.108.175:5060> Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING,PRACK Expires: 300 Event: presence Content-Length: 0 Max-Forwards: 70 I receive the following answer: NOTIFY sip:eugen.dedu@82.238.108.175:5060 SIP/2.0 CSeq: 1 NOTIFY Via: SIP/2.0/UDP 86.64.162.35;branch=z9hG4bK2a99.b8a72c47.0 User-Agent: Kamailio (1.5.3-notls (i386/linux)) From: sip:xyz@ekiga.net;tag=f85b0bd16aaafa8479586ac9f88b3198-10a0 Call-ID: f602824f-1a8e-e011-9efc-0024d693d8e8@snoopy To: sip:eugen.dedu@ekiga.net;tag=4888824f-1a8e-e011-9efc-0024d693d8e8 Contact: <sip:86.64.162.35:5060> Subscription-State: active;expires=370 Event: presence Content-Length: 0 Max-Forwards: 70 To resume: What does SIP standard say about this NOTIFY with empty body? Does this mean that the user xyz is offline? Or does this mean that user's status has not changed? In fact, the NOTIFY with empty body (as shown above) is the first one sent by kamailio, so there is no "previous" state of that user, hence "unchanged" status has no meaning. Thank you, -- Eugen

6 21

Proxy Address
by michel freiha 13 Jun '11

13 Jun '11

Dear All, I'm using kamailio as Proxy server for registration with RTP proxy as relay server for voice packets...I have an issue that when sending an Invite, the audio address in SDP (o header)is not RTP Proxy address while if I send another Invite to kamailio (second Invite), the RTP Proxy address will be present in SDP sent back by Kamailio... Anyone has a clue about what could be that matter? Regards

2 2

Double tuple in notify
by Eugen Dedu 12 Jun '11

12 Jun '11

Hi again, > Then it is the time for Eugen to add it in his sip client and send > the patch for enhancing kamailio's presence server with rfc4481, as > well :o) I start my application, and make my own status be shown. I change my status back and forth a few times between Away and DoNotDisturb (each time a publish is sent), the status returned (in notify) is good. When I change my status to Online/Available (and only to this), kamailio returns a double tuple. Is it normal to have a double tuple? If yes, which one to choose (the one with latest timestamp)? Here it is: NOTIFY sip:eugen.dedu@82.238.108.175:5060 SIP/2.0 CSeq: 7 NOTIFY Via: SIP/2.0/UDP 86.64.162.35;branch=z9hG4bK7634.4d863c14.0 User-Agent: Kamailio (1.5.3-notls (i386/linux)) From: sip:eugen.dedu@ekiga.net;tag=f85b0bd16aaafa8479586ac9f88b3198-29fc Call-ID: 8a0d723b-0f8d-e011-84d7-0024d693d8e8@snoopy To: sip:eugen.dedu@ekiga.net;tag=7c75723b-0f8d-e011-84d7-0024d693d8e8 Contact: <sip:86.64.162.35:5060> Subscription-State: active;expires=270 Event: presence Content-Length: 938 Content-Type: application/pidf+xml Max-Forwards: 70 <?xml version="1.0" encoding="UTF-8"?> <presence xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns: rpid="urn:ietf:params:xml:ns:pidf:rpid" entity="pres:eugen.dedu@ekiga.net"> <tuple id="T00000002"> <status> <basic>open</basic> </status> <contact priority="1">sip:eugen.dedu@ekiga.net</contact> <note>online - </note> <timestamp>2011-06-04T13:55:56+02:00+02:00</timestamp> </tuple> <tuple xmlns="urn:ietf:params:xml:ns:pidf" id="T00000001"> <status> <basic>open</basic> </status> <contact priority="1">sip:eugen.dedu@ekiga.net</contact> <note>away - jjk</note> <timestamp>2011-06-04T13:54:37+02:00+02:00</timestamp> </tuple> <dm:person xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid" id="pid1"> <rpid:activities> <rpid:away/> </rpid:activities> </dm:person> </presence> -- Eugen

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Re: [SR-Users] Meaning of empty body in NOTIFY
by Daniel-Constantin Mierla 10 Jun '11

10 Jun '11

On 6/10/11 10:07 PM, Craig Southeren wrote: > On 10/06/2011 1:01 PM, Daniel-Constantin Mierla wrote:..deleted >>> >>> So, if a client uploads a presence document that contains a >>> <timed-status> element, that element could be sent by the PA when >>> the client goes offline, rather than the entire document being >>> discarded (which is what I suspect is happening now). >> >> we don't have this rfc implemented in kamailio at this time. Is any >> UA you know doing this kind of thing? > > No Then it is the time for Eugen to add it in his sip client and send the patch for enhancing kamailio's presence server with rfc4481, as well ;-) Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

2 1

Functions for DNS
by Nathan Angelacos 10 Jun '11

10 Jun '11

I'm looking to build a set of contacts with q values based on SRV records for serial/parallel forking. I want "enum_query" (loading the contact set with a q value based on the order,preference of the NAPTR record) but the ruri is not an e.164 number, and SRV records are used instead. Think in terms of "find all peer proxies based on SRV lookup, and sort them for use with t_load_contacts()/t_next_contacts()." So far I haven't found any api that exposes the DNS lookups in sip-router. Before rolling something custom with Lua, am I missing something that's already documented? Or are there plans to make such a module? Sure don't want to re-invent the wheel. tia

2 2

New developer: Peter Dunkley
by Daniel-Constantin Mierla 09 Jun '11

09 Jun '11

Hello, just to let everyone to know that starting with today Peter got developer access to GIT repository. He sent many patches in the past months to fix issues or add new features to presence modules (e.g., OMA extensions to embedded XCAP server) -- some are pending to be committed and expect more to come, as he is continuing to work with this part of SIP server. The user id is 'pd'. Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

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Re: [SR-Users] Best way to handle OPTIONS message with Max-Forwards of 0
by Sean O'Donnell 09 Jun '11

09 Jun '11

Looks more comprehensive than mine. Thanks! Sean O'Donnell ---- On Wed, 8 Jun 2011, Klaus Darilion (klaus.mailinglists(a)pernau.at) wrote: Anyway - not standard conform but efficient (put somewhere in the beginning) force_rport(); if (is_method("OPTIONS")) { sl_send_reply("200","OK"); exit; } # sipura/linksys keepalive if (is_method("NOTIFY")) { if ( search("^Event: keep-alive") ) { sl_send_reply("200","Keep Alive"); exit; } } # ignore requests generated by sipvicious # User-Agent: friendly-scanner if ($ua == "friendly-scanner") { exit; } regards Klaus Am 08.06.2011 18:28, schrieb Sean O'Donnell: > Hi: > > We're running kamailio 1.5.5 as part of our VoIP infrastructure. One of our > service providers > has an SBC that sends an OPTIONS messages with a Max-Forwards of 0. > > Looking at RFC 3261 section 11, this seems to be a valid method of forcing the > proxy to respond to the OPTIONS. > However, the standard initial sanity checks in most proxy configs will reject > this message. I fixed our script > by changing to this: > > # initial sanity checks -- messages with > # max_forwards==0, or excessively long requests > if (!mf_process_maxfwd_header("10")) { > if( method != "OPTIONS" ) { > sl_send_reply("483","Too Many Hops"); > return; > }; > }; > if ( msg:len > max_len ) { > sl_send_reply("513", "Message too big"); > return; > }; > > but does anyone have a better solution? > > Thanks, > > Sean O'Donnell > > > _______________________________________________ > 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

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Re: [SR-Users] Best way to handle OPTIONS message with Max-Forwards of 0
by Sean O'Donnell 08 Jun '11

08 Jun '11

Thanks for the reply. In our case, the SBC treated the 483 as an error of some sort and wouldn't send traffic. It wanted a 200/OK. Anyway, I thought that when a proxy responds to an OPTIONS, its supposed to answer with a 200/OK (3261 11.2). Is that correct? Sean O'Donnell ---- On Wed, 8 Jun 2011, Klaus Darilion (klaus.mailinglists(a)pernau.at) wrote: What is the problem with the default config? The proxy will respond with 483 as the standard defines. I guess the SBC will doe this to check if the proxy is alive and will accept any response as "the proxy is alive". regards Klaus Am 08.06.2011 18:28, schrieb Sean O'Donnell: > Hi: > > We're running kamailio 1.5.5 as part of our VoIP infrastructure. One of our > service providers > has an SBC that sends an OPTIONS messages with a Max-Forwards of 0. > > Looking at RFC 3261 section 11, this seems to be a valid method of forcing the > proxy to respond to the OPTIONS. > However, the standard initial sanity checks in most proxy configs will reject > this message. I fixed our script > by changing to this: > > # initial sanity checks -- messages with > # max_forwards==0, or excessively long requests > if (!mf_process_maxfwd_header("10")) { > if( method != "OPTIONS" ) { > sl_send_reply("483","Too Many Hops"); > return; > }; > }; > if ( msg:len > max_len ) { > sl_send_reply("513", "Message too big"); > return; > }; > > but does anyone have a better solution? > > Thanks, > > Sean O'Donnell > > > _______________________________________________ > 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

2 1

Best way to handle OPTIONS message with Max-Forwards of 0
by Sean O'Donnell 08 Jun '11

08 Jun '11

Hi: We're running kamailio 1.5.5 as part of our VoIP infrastructure. One of our service providers has an SBC that sends an OPTIONS messages with a Max-Forwards of 0. Looking at RFC 3261 section 11, this seems to be a valid method of forcing the proxy to respond to the OPTIONS. However, the standard initial sanity checks in most proxy configs will reject this message. I fixed our script by changing to this: # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { if( method != "OPTIONS" ) { sl_send_reply("483","Too Many Hops"); return; }; }; if ( msg:len > max_len ) { sl_send_reply("513", "Message too big"); return; }; but does anyone have a better solution? Thanks, Sean O'Donnell

2 2

Incorrect registrar Expires handling
by Andreas Granig 08 Jun '11

08 Jun '11

Hi, Found an interesting call-flow today where a client (pjsip) is behind NAT and behaves like this during registration (kamailio-3.1.3): REGISTER sip:somedomain.com Contact: <sip:me@10.0.0.7:2001;ob> Expires: 900 SIP/2.0 200 OK Contact: <sip:me@10.0.0.7:2001;ob>;expires=900;received="sip:1.2.3.4:3001" REGISTER sip:somedomain.com Contact: <sip:me@1.2.3.4:3001;transport=UDP;ob> Contact: <sip:me@10.0.0.7:2001;ob>;expires=0 Expires: 900 SIP/2.0 200 OK Contact: <sip:me@10.0.0.7:2001;ob>;expires=900;received="sip:1.2.3.4:3001", <sip:me@1.2.3.4:3001;transport=UDP;ob>;expires=900 So obviously the client is trying to unregister its private contact (for whatever reason - probably to play nice with registrars not supporting far-end NAT traversal), and tries to register the Contact with the "received" address instead. Kamailio however ignores the ";expires=0" for the private Contact and just saves both of them with the expiry value given in the Expires header. Taking the client behavior aside for now: is this the correct behavior of the registrar module? In RFC3261 chapter 10.2.1.1 it says: #+ There are two ways in which a client can suggest an expiration interval for a binding: through an Expires header field or an "expires" Contact header parameter. The latter allows expiration intervals to be suggested on a per-binding basis when more than one binding is given in a single REGISTER request, whereas the former suggests an expiration interval for all Contact header field values that do not contain the "expires" parameter. #- So I'd say that in my specific case "expires" Contact header param (which is 0 for the binding having 10.0.0.7) should de-register this particular binding, whereas the Expires header should be applied to the second binding (having the 1.2.3.4) because it doesn't carry an "expires" param. Opinions on that one? Andreas

2 6

incorrect port 0 in reply from rtp proxy
by Mokhtar Bengana 08 Jun '11

08 Jun '11

I have Kamailio and rtpproxy running behind NAT on the same server. Inbound calls to Kamailio from the outside work fine and I have two way audio but when I try an outbound echo test call the call connects but I have no audio. rtpproxy seems to be running and all rtp ports are poiting to Kamailio internal ip. The syslog is showing this error. Not sure what's missing in the config. Appreciate any inputs. Thanks. root@Kamailio:~# tail /var/log/syslog Jun 3 01:24:02 Kamailio kamailio[810]: INFO: rtpproxy [rtpproxy.c:1403]: rtp proxy <udp:127.0.0.1:22222> found, support for it enabled Jun 3 01:25:05 Kamailio kamailio[791]: INFO: <core> [forward.c:786]: broken reply to forward - no 2nd via Jun 3 01:25:29 Kamailio kamailio[791]: ERROR: rtpproxy [rtpproxy.c:2211]: incorrect port 0 in reply from rtp proxy Jun 3 01:25:30 Kamailio kamailio[793]: ERROR: rtpproxy [rtpproxy.c:2211]: incorrect port 0 in reply from rtp proxy

3 6

t.38 fax
by Sanjeev Singh 07 Jun '11

07 Jun '11

does SER support t.38 fax? what other features related to fax does SER support. please expain in detail. thanks, Sanjeev Singh

2 1

xhttp and dollar sign
by Andreas Granig 06 Jun '11

06 Jun '11

Hi, Is it possible to somehow quote the $ sign in xhttp_send_reply() in order not to be interpreted by kamailio-3.1.3? Tried with "\" and putting the whole body in single quotes, but I always get this error message during startup: ERROR: <core> [pvapi.c:720]: wrong char [$/36] in [${foo}] ERROR: <core> [sr_module.c:1156]: bad PVE format: "${foo}" ERROR: <core> [mod_fix.c:501]: Cannot convert function parameter 1 to spve So I actually just want to return the string "${foo}". Andreas

2 2

Error bad syntax - DEBUG_FLAGS
by Skyler 06 Jun '11

06 Jun '11

Hi all, I'm new to Kamailio (3.1), learning the configuration language. I'm trying to take the advice from one of the docs to create a DEBUG block and place calls everywhere in my script. Right out of the gate I get these errors on loading. Can someone tell me where I went wrong and how I can fix this? It says flag not declared though it's there and I'm a bit lost. 0(10986) : <core> [cfg.y:3412]: parse error in config file //etc/kamailio/kamailio.cfg, line 800, column 6: syntax error 0(10986) : <core> [cfg.y:3412]: parse error in config file //etc/kamailio/kamailio.cfg, line 800, column 6: missing '(' or ')' ? 0(10986) : <core> [cfg.y:3412]: parse error in config file //etc/kamailio/kamailio.cfg, line 800, column 6: bad command: missing ';'? 0(10986) : <core> [cfg.y:3412]: parse error in config file //etc/kamailio/kamailio.cfg, line 800, column 18: '('')' expected (function call) 0(10986) : <core> [cfg.y:3412]: parse error in config file //etc/kamailio/kamailio.cfg, line 800, column 18: bad command: missing ';'? 0(10986) : <core> [cfg.y:3412]: parse error in config file //etc/kamailio/kamailio.cfg, line 800, column 20: bad command 0(10986) : <core> [cfg.y:3412]: parse error in config file //etc/kamailio/kamailio.cfg, line 802, column 24: flag not declared #------------------------------ # Flags #------------------------------ #!define FLAG_DEBUG 1 ... route { # Call debug flags route(ROUTE_DEBUG_FLAGS); ..... route[ROUTE_DEBUG_FLAGS] { resetflag(FLAG_DEBUG); #------------------------------------------------------------ # Methods that will have debugging turned on #----------------------------------------------------------- if (is_method("ACK")) { resetflag(FLAG_DEBUG); } else if (is_method("BYE")) { setflag(FLAG_DEBUG); } else if (is_method("CANCEL")) { setflag(FLAG_DEBUG); } else if (is_method("INVITE")) { setflag(FLAG_DEBUG); } else if (is_method("OPTIONS")) { setflag(FLAG_DEBUG); } else if (is_method("PUBLISH")) { setflag(FLAG_DEBUG); } else if (is_method("REGISTER")) { resetflag(FLAG_DEBUG); } else if (is_method("MESSAGE")) { setflag(FLAG_DEBUG); } else if (is_method("NOTIFY")) { setflag(FLAG_DEBUG); } else if (is_method("SUBSCRIBE")) { setflag(FLAG_DEBUG); } else if (is_method("INFO")) { setflag(FLAG_DEBUG); } else if (is_method("PRACK")) { setflag(FLAG_DEBUG); } else if (is_method("REFER")) { setflag(FLAG_DEBUG); } else { setflag(FLAG_DEBUG); } } }

2 2

Using =~ to get the group identifier with longest match
by Iñaki Baz Castillo 06 Jun '11

06 Jun '11

Hi, I wonder how feasible is the folowing: I have a regular expresion like: REGEXP = "^((1|20)|(13|3|4)|(5|6))" There are 3 matching groups: 1) (1|20) 2) (13|3|4) 3) (5|6) I want to match the RURI user, let's suppose $rU = "138787686223". In this case the above regexp matches groups 1 and 2, but since group 2 matching is longest ("13" is longest than "1") I want to get value 2. In case $rU = "999999" I need to get "null", in case $rU = "1000022323" I need to get "1". AFAIK there is no way to do it with current regular expression utils in Kamailio, but maybe I miss something. Thanks a lot. -- Iñaki Baz Castillo <ibc(a)aliax.net>

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Questino about dialplan module
by Gary Chen 06 Jun '11

06 Jun '11

Kamailio 3.1.2 I am testing out dialplan module. Every time when I issue 'kamctl dialplan show', it shows what is in the database even if I have not yet issue "kamctl dialplan reload' after I changed data in database. I thought that dialplan is stored in the memory and you have to issue the reload before you can see the change. Why the ' kamctl dialplan show' display the data directly from mysql database? Does that mean that dialplan data is not stored in the memory? Gary

3 2

callback function of tm module
by Bruno Bresciani 06 Jun '11

06 Jun '11

Hi All, I'm having doubts in the implementation of the callback function module tm. As she calls the function to record register_tmcb ()? Can someome help me? Regards

2 5

Re: [SR-Users] Questino about dialplan module
by Gary Chen 03 Jun '11

03 Jun '11

I am talking about kamctl dialplan command not mi. When you type in kamctl, it lists a lot commands. Are these commands still valid? See below: kamctl ../../sbin/kamctl 3.1.0 Existing commands: -- command 'start|stop|restart' restart ............................ restart Kamailio start .............................. start Kamailio stop ............................... stop Kamailio -- command 'acl' - manage access control lists (acl) acl show [<username>] .............. show user membership acl grant <username> <group> ....... grant user membership (*) acl revoke <username> [<group>] .... grant user membership(s) (*) -- command 'lcr' - manage least cost routes (lcr) lcr show_gws....... show database gateways lcr show_routes.... show database routes lcr dump_gws....... show in memory gateways lcr dump_routes.... show in memory routes lcr reload ........ reload lcr gateways and routes lcr eval_weights .. evaluates probability for given GW's weights -- command 'cr' - manage carrierroute tables cr show ....................................................... show tables cr reload ..................................................... reload tables cr dump ....................................................... show in memory tables cr addcn <carrier id> <carrier name> .......................... add a carrier name cr rmcn <carrier id> ......................................... rm a carrier name cr adddn <domain id> <domain name> ............................ add a domain name cr rmdn <domain id> .......................................... rm a domain name cr addcarrier <carrier> <scan_prefix> <domain> <rewrite_host> ................ <prob> <strip> <rewrite_prefix> <rewrite_suffix> ............... <flags> <mask> <comment> .........................add a carrier (prob, strip, rewrite_prefix, rewrite_suffix,................... flags, mask and comment are optional arguments) ............... cr rmcarrier <carrier> <scan_prefix> <domain> ................ rm a carrier -- command 'rpid' - manage Remote-Party-ID (RPID) rpid add <username> <rpid> ......... add rpid for a user (*) rpid rm <username> ................. set rpid to NULL for a user (*) rpid show <username> ............... show rpid of a user -- command 'add|passwd|rm' - manage subscribers add <username> <password> .......... add a new subscriber (*) passwd <username> <passwd> ......... change user's password (*) rm <username> ...................... delete a user (*) -- command 'add|dump|reload|rm|show' - manage trusted trusted show ...................... show db content trusted dump ...................... show cache content trusted reload .................... reload db table into cache trusted add <src_ip> <proto> <from_pattern> <tag> ....................... add a new entry ....................... (from_pattern and tag are optional arguments) trusted rm <src_ip> ............... remove all entres for the given src_ip -- command 'dispatcher' - manage dispatcher * Examples: dispatcher addgw 1 sip:1.2.3.1:5050 1 'outbound gateway' * dispatcher addgw 2 sip:1.2.3.4:5050 3 '' * dispatcher rmgw 4 dispatcher show ..................... show dispatcher gateways dispatcher reload ................... reload dispatcher gateways dispatcher dump ..................... show in memory dispatcher gateways dispatcher addgw <setid> <destination> <flags> <description> .......................... add gateway dispatcher rmgw <id> ................ delete gateway -- command 'db' - database operations db exec <query> ..................... execute SQL query db roexec <roquery> ................. execute read-only SQL query db run <id> ......................... execute SQL query from $id variable db rorun <id> ....................... execute read-only SQL query from $id variable db show <table> ..................... display table content -- command 'speeddial' - manage speed dials (short numbers) speeddial show <speeddial-id> ....... show speeddial details speeddial list <sip-id> ............. list speeddial for uri speeddial add <sip-id> <sd-id> <new-uri> [<desc>] ... ........................... add a speedial (*) speeddial rm <sip-id> <sd-id> ....... remove a speeddial (*) speeddial help ...................... help message - <speeddial-id>, <sd-id> must be an AoR (username@domain) - <sip-id> must be an AoR (username@domain) - <new-uri> must be a SIP AoR (sip:username@domain) - <desc> a description for speeddial -- command 'avp' - manage AVPs avp list [-T table] [-u <sip-id|uuid>] [-a attribute] [-v value] [-t type] ... list AVPs avp add [-T table] <sip-id|uuid> <attribute> <type> <value> ............ add AVP (*) avp rm [-T table] [-u <sip-id|uuid>] [-a attribute] [-v value] [-t type] ... remove AVP (*) avp help .................................. help message - -T - table name - -u - SIP id or unique id - -a - AVP name - -v - AVP value - -t - AVP name and type (0 (str:str), 1 (str:int), 2 (int:str), 3 (int:int)) - <sip-id> must be an AoR (username@domain) - <uuid> must be a string but not AoR -- command 'alias_db' - manage database aliases alias_db show <alias> .............. show alias details alias_db list <sip-id> ............. list aliases for uri alias_db add <alias> <sip-id> ...... add an alias (*) alias_db rm <alias> ................ remove an alias (*) alias_db help ...................... help message - <alias> must be an AoR (username@domain)" - <sip-id> must be an AoR (username@domain)" -- command 'domain' - manage local domains domain reload ....................... reload domains from disk domain show ......................... show current domains in memory domain showdb ....................... show domains in the database domain add <domain> ................. add the domain to the database domain rm <domain> .................. delete the domain from the database -- command 'cisco_restart' - restart CISCO phone (NOTIFY) cisco_restart <uri> ................ restart phone configured for <uri> -- command 'online' - dump online users from memory online ............................. display online users -- command 'monitor' - show internal status monitor ............................ show server's internal status -- command 'ping' - ping a SIP URI (OPTIONS) ping <uri> ......................... ping <uri> with SIP OPTIONS -- command 'ul|alias' - manage user location or aliases ul show [<username>]................ show in-RAM online users ul show --brief..................... show in-RAM online users in short format ul rm <username> [<contact URI>].... delete user's usrloc entries ul add <username> <uri> ............ introduce a permanent usrloc entry ul add <username> <uri> <expires> .. introduce a temporary usrloc entry -- command 'fifo' fifo ............................... send raw FIFO command -- command 'cisco_restart' - restart CISCO phone (NOTIFY) cisco_restart <uri> ................ restart phone configured for <uri> -- command 'online' - dump online users from memory online ............................. display online users -- command 'monitor' - show internal status monitor ............................ show server's internal status -- command 'ping' - ping a SIP URI (OPTIONS) ping <uri> ......................... ping <uri> with SIP OPTIONS -- command 'ul|alias' - manage user location or aliases ul show [<username>]................ show in-RAM online users ul show --brief..................... show in-RAM online users in short format ul rm <username> [<contact URI>].... delete user's usrloc entries ul add <username> <uri> ............ introduce a permanent usrloc entry ul add <username> <uri> <expires> .. introduce a temporary usrloc entry -----Original Message----- From: sr-users-bounces(a)lists.sip-router.org [mailto:sr-users-bounces@lists.sip-router.org] On Behalf Of Juha Heinanen Sent: Friday, June 03, 2011 3:50 PM To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) -Users Mailing List Subject: [SR-Users] Questino about dialplan module Gary Chen writes: > I am testing out dialplan module. Every time when I issue 'kamctl > dialplan show', it shows what is in the database even if I have not > yet issue "kamctl dialplan reload' after I changed data in database. I > thought that dialplan is stored in the memory and you have to issue > the reload before you can see the change. Why the ' kamctl dialplan > show' display the data directly from mysql database? Does that mean > that dialplan data is not stored in the memory? there is no mi function to show dialplan rules in memory. i have no idea, what 'show' does. please read readme of dialplan module before asking questions about the module. -- juha _______________________________________________ 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

IRC meeting today, 15:00GMT
by Daniel-Constantin Mierla 03 Jun '11

03 Jun '11

Hello, everyone that wants to discuss about the next major release (v3.2.0), please join us on IRC: Channel: #sip-router Server: irc.freenode.net Time: 15:00GMT/London (16:00 Berlin/Paris/Madrid, 10:00am New York) Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 1

IMC module - #join to private room crashes kamailio
by Ozren Lapcevic 03 Jun '11

03 Jun '11

Hi, there is a bug in IMC module that crashes Kamailio. Here is the scenario: One user (pero(a)abc.hr) creates private chat room ( chat-999(a)abc.hr). Other user (oz(a)abc.hr) tries to join the chat-999(a)abc.hrby sending #join. Kamailio crashes at that point. Here is the log: Best regards, Ozren Jun 1 15:10:37 ser-test kernel: [5844312.456709] kamailio[20118]: segfault at 10 ip 7f28f89dae8e sp 7fffffffb750 error 4 in imc.so[7f28f89cf000+17000] Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20118]: INFO: <script>: going to imc: F [sip:oz@abc.hr] T [sip:chat-999@abc.hr] B [#join] Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20121]: DEBUG: <core> [receive.c:186]: preparing to run routing scripts... Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20118]: DEBUG: imc [imc.c:525]: found command Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20121]: DEBUG: maxfwd [mf_funcs.c:85]: value = 70 Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20118]: DEBUG: imc [imc_cmd.c:146]: command: [join] Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20121]: DEBUG: <core> [parser/msg_parser.c:201]: DEBUG: get_hdr_body : content_length=223 Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20118]: DEBUG: imc [imc_cmd.c:292]: found room [chat-999] Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20121]: DEBUG: <core> [parser/msg_parser.c:103]: found end of header Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20118]: ERROR: imc [imc_cmd.c:317]: attept to join private room [chat-999] from user [oz] Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20121]: DEBUG: <core> [parser/parse_to.c:174]: DEBUG: add_param: tag=5e3288a6 Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20113]: ALERT: <core> [main.c:741]: child process 20118 exited by a signal 11 Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20121]: DEBUG: <core> [parser/parse_to.c:803]: end of header reached, state=29 Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20119]: DEBUG: <core> [parser/msg_parser.c:630]: SIP Request: Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20113]: ALERT: <core> [main.c:744]: core was not generated Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20121]: DEBUG: <core> [msg_translator.c:204]: check_via_address(161.53.255.82, 161.53.255.82, 0) Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20119]: DEBUG: <core> [parser/msg_parser.c:632]: method: <MESSAGE> Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20113]: INFO: <core> [main.c:756]: INFO: terminating due to SIGCHLD Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20114]: INFO: <core> [main.c:807]: INFO: signal 15 received Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20115]: INFO: <core> [main.c:807]: INFO: signal 15 received Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20116]: INFO: <core> [main.c:807]: INFO: signal 15 received Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20117]: INFO: <core> [main.c:807]: INFO: signal 15 received Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20122]: INFO: <core> [main.c:807]: INFO: signal 15 received Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20124]: INFO: <core> [main.c:807]: INFO: signal 15 received Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20125]: INFO: <core> [main.c:807]: INFO: signal 15 received Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20129]: INFO: <core> [main.c:807]: INFO: signal 15 received Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20128]: INFO: <core> [main.c:807]: INFO: signal 15 received Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20127]: INFO: <core> [main.c:807]: INFO: signal 15 received Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20123]: INFO: <core> [main.c:807]: INFO: signal 15 received Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20126]: INFO: <core> [main.c:807]: INFO: signal 15 received Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20120]: INFO: <core> [main.c:807]: INFO: signal 15 received Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20130]: INFO: <core> [main.c:807]: INFO: signal 15 received Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20131]: INFO: <core> [main.c:807]: INFO: signal 15 received Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20113]: DEBUG: imc [imc.c:709]: room 3 chat-777 Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20113]: DEBUG: imc [imc.c:709]: room 11 chat-888 Jun 1 15:10:37 ser-test /usr/local/kamailio-ozren/sbin/kamailio[20113]: DEBUG: imc [imc.c:709]: room 13 chat-999

2 6

perlvdb error
by Roman Yeryomin 03 Jun '11

03 Jun '11

On 18 April 2011 15:47, Henning Westerholt <henning.westerholt(a)1und1.de> wrote: > On Monday 18 April 2011, Roman Yeryomin wrote: >> coredump? easy :) >> you can pick core file itself here (it's pretty heavy to send to the list): >> http://83.99.159.113/core.kamailio.20110418-1331 >> >> What do you mean by backtrace? log output or strace output? > > Hi Roman, > > he refers to a gdb backtrace, "gdb bt". More informations e.g. here: > http://www.kamailio.org/dokuwiki/doku.php/troubleshooting:corefiles > ok, here is the backtrace: # gdb kamailio core GNU gdb 6.8-debian Copyright (C) 2008 Free Software Foundation, Inc. License GPLv3+: GNU GPL version 3 or later <http://gnu.org/licenses/gpl.html> This is free software: you are free to change and redistribute it. There is NO WARRANTY, to the extent permitted by law. Type "show copying" and "show warranty" for details. This GDB was configured as "i486-linux-gnu"... warning: Can't read pathname for load map: Input/output error. Reading symbols from /lib/tls/i686/cmov/libdl.so.2...done. Loaded symbols for /lib/tls/i686/cmov/libdl.so.2 Reading symbols from /lib/tls/i686/cmov/libresolv.so.2...done. Loaded symbols for /lib/tls/i686/cmov/libresolv.so.2 Reading symbols from /lib/tls/i686/cmov/libc.so.6...done. Loaded symbols for /lib/tls/i686/cmov/libc.so.6 Reading symbols from /lib/ld-linux.so.2...done. Loaded symbols for /lib/ld-linux.so.2 Reading symbols from /usr/local/lib/kamailio/modules_k/mi_fifo.so...done. Loaded symbols for /usr/local/lib/kamailio/modules_k/mi_fifo.so Reading symbols from /usr/local/lib/kamailio/libkmi.so.1...done. Loaded symbols for /usr/local/lib/kamailio/libkmi.so.1 Reading symbols from /usr/local/lib/kamailio/modules_k/kex.so...done. Loaded symbols for /usr/local/lib/kamailio/modules_k/kex.so Reading symbols from /usr/local/lib/kamailio/libkcore.so.1...done. Loaded symbols for /usr/local/lib/kamailio/libkcore.so.1 Reading symbols from /usr/local/lib/kamailio/modules/tm.so...done. Loaded symbols for /usr/local/lib/kamailio/modules/tm.so Reading symbols from /usr/local/lib/kamailio/modules_k/tmx.so...done. Loaded symbols for /usr/local/lib/kamailio/modules_k/tmx.so Reading symbols from /usr/local/lib/kamailio/modules/sl.so...done. Loaded symbols for /usr/local/lib/kamailio/modules/sl.so Reading symbols from /usr/local/lib/kamailio/modules_k/rr.so...done. Loaded symbols for /usr/local/lib/kamailio/modules_k/rr.so Reading symbols from /usr/local/lib/kamailio/modules_k/pv.so...done. Loaded symbols for /usr/local/lib/kamailio/modules_k/pv.so Reading symbols from /usr/local/lib/kamailio/modules_k/maxfwd.so...done. Loaded symbols for /usr/local/lib/kamailio/modules_k/maxfwd.so Reading symbols from /usr/local/lib/kamailio/modules_k/usrloc.so...done. Loaded symbols for /usr/local/lib/kamailio/modules_k/usrloc.so Reading symbols from /usr/local/lib/kamailio/libsrdb1.so.1...done. Loaded symbols for /usr/local/lib/kamailio/libsrdb1.so.1 Reading symbols from /usr/local/lib/kamailio/modules_k/registrar.so...done. Loaded symbols for /usr/local/lib/kamailio/modules_k/registrar.so Reading symbols from /usr/local/lib/kamailio/modules_k/textops.so...done. Loaded symbols for /usr/local/lib/kamailio/modules_k/textops.so Reading symbols from /usr/local/lib/kamailio/modules_k/siputils.so...done. Loaded symbols for /usr/local/lib/kamailio/modules_k/siputils.so Reading symbols from /usr/local/lib/kamailio/modules_k/xlog.so...done. Loaded symbols for /usr/local/lib/kamailio/modules_k/xlog.so Reading symbols from /usr/local/lib/kamailio/modules/sanity.so...done. Loaded symbols for /usr/local/lib/kamailio/modules/sanity.so Reading symbols from /usr/local/lib/kamailio/modules/ctl.so...done. Loaded symbols for /usr/local/lib/kamailio/modules/ctl.so Reading symbols from /usr/local/lib/kamailio/modules/mi_rpc.so...done. Loaded symbols for /usr/local/lib/kamailio/modules/mi_rpc.so Reading symbols from /usr/local/lib/kamailio/modules_k/acc.so...done. Loaded symbols for /usr/local/lib/kamailio/modules_k/acc.so Reading symbols from /lib/tls/i686/cmov/libpthread.so.0...done. Loaded symbols for /lib/tls/i686/cmov/libpthread.so.0 Reading symbols from /usr/local/lib/kamailio/modules_k/perl.so...done. Loaded symbols for /usr/local/lib/kamailio/modules_k/perl.so Reading symbols from /usr/lib/libperl.so.5.8...done. Loaded symbols for /usr/lib/libperl.so.5.8 Reading symbols from /lib/tls/i686/cmov/libm.so.6...done. Loaded symbols for /lib/tls/i686/cmov/libm.so.6 Reading symbols from /lib/tls/i686/cmov/libcrypt.so.1...done. Loaded symbols for /lib/tls/i686/cmov/libcrypt.so.1 Reading symbols from /usr/local/lib/kamailio/modules_k/perlvdb.so...done. Loaded symbols for /usr/local/lib/kamailio/modules_k/perlvdb.so Reading symbols from /usr/local/lib/kamailio/modules/auth.so...done. Loaded symbols for /usr/local/lib/kamailio/modules/auth.so Reading symbols from /usr/local/lib/kamailio/modules_k/auth_db.so...done. Loaded symbols for /usr/local/lib/kamailio/modules_k/auth_db.so Reading symbols from /usr/local/lib/kamailio/modules/app_lua.so...done. Loaded symbols for /usr/local/lib/kamailio/modules/app_lua.so Reading symbols from /usr/lib/liblua5.1.so.0...done. Loaded symbols for /usr/lib/liblua5.1.so.0 Reading symbols from /lib/tls/i686/cmov/libnss_files.so.2...done. Loaded symbols for /lib/tls/i686/cmov/libnss_files.so.2 Reading symbols from /lib/tls/i686/cmov/libnss_dns.so.2...done. Loaded symbols for /lib/tls/i686/cmov/libnss_dns.so.2 Core was generated by `kamailio -dddd'. Program terminated with signal 11, Segmentation fault. [New process 13773] #0 0xb7228e3a in parseurl (url=0xb7213230 "T�0\b#") at perlvdbfunc.c:60 60 if (strlen(cn) > 0) If I understand this correctly error refers to row 60 in perlvdbfunc.c strlen expects const char* instead of char* ? Regards, Roman

2 17

Re: [SR-Users] Problem using dispatcher.
by Gary Chen 02 Jun '11

02 Jun '11

That works. Thanks. From: Daniel-Constantin Mierla [mailto:miconda@gmail.com] Sent: Thursday, June 02, 2011 4:28 PM To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List Cc: Gary Chen Subject: Re: [SR-Users] Problem using dispatcher. On 6/2/11 9:41 PM, Gary Chen wrote: Thanks for the quick reply. How can I fix this? When you set t_on_failure for an INVITE, the respective failure route will be executed for any reply >=0. What you have to do is to not re-route to a new destination from failure route, simply exit for those reply codes, like: if(t_check_status("401|407")) exit; You don't handle as well the canceled transactions, when re-routing makes no sense -- you should do it (see default config file for example. Cheers, Daniel We are using Kamailio as single point of entry for all our phones. Our phones all registered with Freeswitch. Kamailio only route the SIP traffic. Is there any way I can make Kamailio not treat 407 as failure? I also have the same problem with REGISTER messages when Freeswitch reply with '401 Unauthorized'. I end up using Forward() instead of t_relay() to make the phone1 register with Freeswitch. Any suggestion? -----Original Message----- From: sr-users-bounces(a)lists.sip-router.org [ mailto:sr-users-bounces@lists.sip-router.org ] On Behalf Of Alex Balashov Sent: Thursday, June 02, 2011 3:30 PM To: sr-users(a)lists.sip-router.org Subject: Re: [SR-Users] Problem using dispatcher. On 06/02/2011 03:26 PM, Gary Chen wrote: > the dispatcher always get into FAILURE route although the log shows > that kamailio did send INVITE to 10.10.1.222 and received '407 - > Proxy Authentication Required' from Freexwitch1 and then dispatcher > send INVITE to 10.10.1.223 and also received '407 - Proxy > Authentication Required' from Freexwitch2 and eventually failed. That's because any reply >= 300 is considered a failure. -- 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 _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

Re: [SR-Users] Problem using dispatcher.
by Gary Chen 02 Jun '11

02 Jun '11

Thanks for the quick reply. How can I fix this? We are using Kamailio as single point of entry for all our phones. Our phones all registered with Freeswitch. Kamailio only route the SIP traffic. Is there any way I can make Kamailio not treat 407 as failure? I also have the same problem with REGISTER messages when Freeswitch reply with '401 Unauthorized'. I end up using Forward() instead of t_relay() to make the phone1 register with Freeswitch. Any suggestion? -----Original Message----- From: sr-users-bounces(a)lists.sip-router.org [mailto:sr-users-bounces@lists.sip-router.org] On Behalf Of Alex Balashov Sent: Thursday, June 02, 2011 3:30 PM To: sr-users(a)lists.sip-router.org Subject: Re: [SR-Users] Problem using dispatcher. On 06/02/2011 03:26 PM, Gary Chen wrote: > the dispatcher always get into FAILURE route although the log shows > that kamailio did send INVITE to 10.10.1.222 and received '407 - > Proxy Authentication Required' from Freexwitch1 and then dispatcher > send INVITE to 10.10.1.223 and also received '407 - Proxy > Authentication Required' from Freexwitch2 and eventually failed. That's because any reply >= 300 is considered a failure. -- 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 1

Problem using dispatcher.
by Gary Chen 02 Jun '11

02 Jun '11

Kamailio 3.1.2 Here is my setup: Phone1 (10.10.1.114) | Kamailio (10.10.1.230) / \ / \ Freeswitch1 Freeswitch2 ----------------------> Other phones (10.10.1.222) (10.10.1.223) Both Freeswitch's share the same REGISTRATION information. Phone1 is already registered with Freeswitch. When I try to dial 'Other phones' using phone1, the dispatcher always get into FAILURE route although the log shows that kamailio did send INVITE to 10.10.1.222 and received '407 - Proxy Authentication Required' from Freexwitch1 and then dispatcher send INVITE to 10.10.1.223 and also received '407 - Proxy Authentication Required' from Freexwitch2 and eventually failed. Any idea? Here is my configuration: modparam("dispatcher", "db_url", DBURL) modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "ds_ping_interval", 10000) modparam("dispatcher", "ds_probing_threshhold", 3) #modparam("dispatcher", "ds_ping_reply_codes", "class=2;class=4") modparam("dispatcher", "ds_ping_method", "OPTIONS") modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_ping_from", "sip:proxy@SIP-router1.testnet.net") modparam("dispatcher", "dst_avp", "$avp(dsdst)") modparam("dispatcher", "grp_avp", "$avp(dsgrp)") modparam("dispatcher", "cnt_avp", "$avp(dscnt)") modparam("dispatcher", "attrs_avp", "$avp(dsattrs)") modparam("dispatcher", "flags", 3) modparam("dispatcher", "force_dst", 1) route { .... if ( ds_select_dst("$var(group)", "9") ){ route(RELAY_DS); return; ... } route[[RELAY_DS] if (is_method("INVITE")) { t_on_branch("BRANCH_ONE"); t_on_reply("NAT_REPLY"); t_on_failure("FAIL_DS"); } if (!t_relay()) { sl_reply_error(); } } failure_route[FAIL_DS] { if (t_check_status("408")) { ds_mark_dst("i"); } if (!ds_next_dst()) { t_reply("503", "Service unavailable: no more dst"); exit; } t_on_failure("FAIL_DS"); route(RELAY_DS); } onreply_route[NAT_REPLY] { xlog("L_INFO", "REPLY_ROUTE: S=$rs - $rr F=$fu T=$tu IP=$avp(s:ip):$avp(s:port) ($si:$sp) ID=$ci\n"); } And here is the SIP messages from ngrep: U 10.10.1.144:51029 -> 10.10.1.230:5060 INVITE sip:1001@fs2000.testnet.net;user=phone SIP/2.0..Via: SIP/2.0/UDP 10.10.1.144:5060;branch=z9hG4bK116d554d..F rom: "Line1" <sip:5022441011@fs2000.testnet.net>;tag=00082166efcb002153bc0a31-20004ffd..To: <sip:1001@fs2000.testnet .net;user=phone>..Call-ID: 00082166-efcb0013-692f44d9-6760a891@10.10.1.144..Max-Forwards: 70..Date: Thu, 02 Jun 2011 19:14:14 GMT..CSeq: 101 INVITE..User-Agent: Cisco-CP7960G/7.5..Contact: <sip:5022441011@10.10.1.144:5060>..Expires: 180..Accept: application/sdp..Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE..Supported: replaces..C ontent-Length: 261..Content-Type: application/sdp..Content-Disposition: session;handling=optional....v=0..o=Cisco-SIPUA 10962 0 IN IP4 10.10.1.144..s=SIP Call..t=0 0..m=audio 20372 RTP/AVP 0 8 18 101..c=IN IP4 10.10.1.144..a=rtpmap: 0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:18 G729/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=sendrecv .. # U 10.10.1.230:5060 -> 10.10.1.144:5060 SIP/2.0 100 trying -- your call is important to us..Via: SIP/2.0/UDP 10.10.1.144:5060;branch=z9hG4bK116d554d..From: "Line1" <sip:5022441011@fs2000.testnet.net>;tag=00082166efcb002153bc0a31-20004ffd..To: <sip:1001@fs2000.testnet.net; user=phone>..Call-ID: 00082166-efcb0013-692f44d9-6760a891@10.10.1.144..CSeq: 101 INVITE..Server: kamailio (3.1.2 (x8 6_64/linux))..Content-Length: 0.... # U 10.10.1.230:5060 -> 10.10.1.222:5060 INVITE sip:1001@fs2000.testnet.net;user=phone SIP/2.0..Record-Route: <sip:10.10.1.230;lr=on;nat=yes>..Via: SIP/2.0/ UDP 10.10.1.230;branch=z9hG4bK1ef6.c9d52d72.0..Via: SIP/2.0/UDP 10.10.1.144:5060;branch=z9hG4bK116d554d..From: "L ine1" <sip:5022441011@fs2000.testnet.net>;tag=00082166efcb002153bc0a31-20004ffd..To: <sip:1001@fs2000.testnet.net;us er=phone>..Call-ID: 00082166-efcb0013-692f44d9-6760a891@10.10.1.144..Max-Forwards: 69..Date: Thu, 02 Jun 2011 19:14: 14 GMT..CSeq: 101 INVITE..User-Agent: Cisco-CP7960G/7.5..Contact: <sip:5022441011@10.10.1.144:5060>..Expires: 180..A ccept: application/sdp..Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE..Supported: replaces..Content- Length: 277..Content-Type: application/sdp..Content-Disposition: session;handling=optional....v=0..o=Cisco-SIPUA 10962 0 IN IP4 10.10.1.231..s=SIP Call..t=0 0..m=audio 38426 RTP/AVP 0 8 18 101..c=IN IP4 10.10.1.231..a=rtpmap:0 PCMU/80 00..a=rtpmap:8 PCMA/8000..a=rtpmap:18 G729/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=sendrecv..a=nortp proxy:yes.. # U 10.10.1.222:5060 -> 10.10.1.230:5060 SIP/2.0 100 Trying..Via: SIP/2.0/UDP 10.10.1.230;branch=z9hG4bK1ef6.c9d52d72.0..Via: SIP/2.0/UDP 10.10.1.144:5060 ;branch=z9hG4bK116d554d..Record-Route: <sip:10.10.1.230;lr=on;nat=yes>..From: "Line1" <sip:5022441011@fs2000.lightyea r.net>;tag=00082166efcb002153bc0a31-20004ffd..To: <sip:1001@fs2000.testnet.net;user=phone>..Call-ID: 00082166-efcb0013 -692f44d9-6760a891@10.10.1.144..CSeq: 101 INVITE..User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-f17e962 2011-04-19 0 3-05-32 -0400..Content-Length: 0.... # U 10.10.1.222:5060 -> 10.10.1.230:5060 SIP/2.0 407 Proxy Authentication Required..Via: SIP/2.0/UDP 10.10.1.230;branch=z9hG4bK1ef6.c9d52d72.0..Via: SIP/2.0/U DP 10.10.1.144:5060;branch=z9hG4bK116d554d..From: "Line1" <sip:5022441011@fs2000.testnet.net>;tag=00082166efcb0021 53bc0a31-20004ffd..To: <sip:1001@fs2000.testnet.net;user=phone>;tag=6mjQm2eayZ49j..Call-ID: 00082166-efcb0013-692f44d9 -6760a891@10.10.1.144..CSeq: 101 INVITE..User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-f17e962 2011-04-19 03-05-32 - 0400..Accept: application/sdp..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE..Supported: timer, precondition, path, replaces..Allow-Events: talk, hold, presence, dialog, line-se ize, call-info, sla, include-session-description, presence.winfo, message-summary, refer..Proxy-Authenticate: Digest rea lm="fs2000.testnet.net", nonce="8da1ddb1-e393-442c-8be0-17e439457363", algorithm=MD5, qop="auth"..Content-Length: 0... . # U 10.10.1.230:5060 -> 10.10.1.222:5060 ACK sip:1001@fs2000.testnet.net;user=phone SIP/2.0..Via: SIP/2.0/UDP 10.10.1.230;branch=z9hG4bK1ef6.c9d52d72.0..Fro m: "Line1" <sip:5022441011@fs2000.testnet.net>;tag=00082166efcb002153bc0a31-20004ffd..To: <sip:1001@fs2000.testnet.n et;user=phone>;tag=6mjQm2eayZ49j..Call-ID: 00082166-efcb0013-692f44d9-6760a891@10.10.1.144..Max-Forwards: 69..CSeq: 101 ACK..Content-Length: 0.... # U 10.10.1.230:5060 -> 10.10.1.223:5060 INVITE sip:1001@fs2000.testnet.net;user=phone SIP/2.0..Record-Route: <sip:10.10.1.230;lr=on;nat=yes;nat=yes>..Via: SIP/2.0/UDP 10.10.1.230;branch=z9hG4bK1ef6.c9d52d72.1..Via: SIP/2.0/UDP 10.10.1.144:5060;branch=z9hG4bK116d554d.. From: "Line1" <sip:5022441011@fs2000.testnet.net>;tag=00082166efcb002153bc0a31-20004ffd..To: <sip:1001@fs2000.lightyea r.net;user=phone>..Call-ID: 00082166-efcb0013-692f44d9-6760a891@10.10.1.144..Max-Forwards: 69..Date: Thu, 02 Jun 201 1 19:14:14 GMT..CSeq: 101 INVITE..User-Agent: Cisco-CP7960G/7.5..Contact: <sip:5022441011@10.10.1.144:5060>..Expires : 180..Accept: application/sdp..Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE..Supported: replaces.. Content-Length: 328..Content-Type: application/sdp..Content-Disposition: session;handling=optional....v=0..o=Cisco-SIPUA 10962 0 IN IP4 10.10.1.23110.10.1.231..s=SIP Call..t=0 0..m=audio 3842638426 RTP/AVP 0 8 18 101..c=IN IP4 216.49. 129.23110.10.1.231..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:18 G729/8000..a=rtpmap:101 telephone-event/8 000..a=fmtp:101 0-15..a=sendrecv..a=nortpproxy:yes..a=nortpproxy:yes.. # U 10.10.1.223:5060 -> 10.10.1.230:5060 SIP/2.0 100 Trying..Via: SIP/2.0/UDP 10.10.1.230;branch=z9hG4bK1ef6.c9d52d72.1..Via: SIP/2.0/UDP 10.10.1.144:5060 ;branch=z9hG4bK116d554d..Record-Route: <sip:10.10.1.230;lr=on;nat=yes;nat=yes>..From: "Line1" <sip:5022441011@fs2000. testnet.net>;tag=00082166efcb002153bc0a31-20004ffd..To: <sip:1001@fs2000.testnet.net;user=phone>..Call-ID: 00082166- efcb0013-692f44d9-6760a891@10.10.1.144..CSeq: 101 INVITE..User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-f17e962 2011 -04-19 03-05-32 -0400..Content-Length: 0.... # U 10.10.1.223:5060 -> 10.10.1.230:5060 SIP/2.0 407 Proxy Authentication Required..Via: SIP/2.0/UDP 10.10.1.230;branch=z9hG4bK1ef6.c9d52d72.1..Via: SIP/2.0/U DP 10.10.1.144:5060;branch=z9hG4bK116d554d..From: "Line1" <sip:5022441011@fs2000.testnet.net>;tag=00082166efcb0021 53bc0a31-20004ffd..To: <sip:1001@fs2000.testnet.net;user=phone>;tag=Ky0XB4gDeDy6F..Call-ID: 00082166-efcb0013-692f44d9 -6760a891@10.10.1.144..CSeq: 101 INVITE..User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-f17e962 2011-04-19 03-05-32 - 0400..Accept: application/sdp..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE..Supported: timer, precondition, path, replaces..Allow-Events: talk, hold, presence, dialog, line-se ize, call-info, sla, include-session-description, presence.winfo, message-summary, refer..Proxy-Authenticate: Digest rea lm="fs2000.testnet.net", nonce="8769298b-c52d-43b8-b69c-6ebf26f3184d", algorithm=MD5, qop="auth"..Content-Length: 0... . # U 10.10.1.230:5060 -> 10.10.1.223:5060 ACK sip:1001@fs2000.testnet.net;user=phone SIP/2.0..Via: SIP/2.0/UDP 10.10.1.230;branch=z9hG4bK1ef6.c9d52d72.1..Fro m: "Line1" <sip:5022441011@fs2000.testnet.net>;tag=00082166efcb002153bc0a31-20004ffd..To: <sip:1001@fs2000.testnet.n et;user=phone>;tag=Ky0XB4gDeDy6F..Call-ID: 00082166-efcb0013-692f44d9-6760a891@10.10.1.144..Max-Forwards: 69..CSeq: 101 ACK..Content-Length: 0.... # U 10.10.1.230:5060 -> 10.10.1.144:5060 SIP/2.0 503 Service unavailable: no more dst..Via: SIP/2.0/UDP 10.10.1.144:5060;branch=z9hG4bK116d554d..From: "Line1 " <sip:5022441011@fs2000.testnet.net>;tag=00082166efcb002153bc0a31-20004ffd..To: <sip:1001@fs2000.testnet.net;user=p hone>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-2c94..Call-ID: 00082166-efcb0013-692f44d9-6760a891@10.10.1.144..CSeq: 101 INVITE..Server: kamailio (3.1.2 (x86_64/linux))..Content-Length: 0.... # U 10.10.1.144:51030 -> 10.10.1.230:5060 ACK sip:1001@fs2000.testnet.net;user=phone SIP/2.0..Via: SIP/2.0/UDP 10.10.1.144:5060;branch=z9hG4bK116d554d..From : "Line1" <sip:5022441011@fs2000.testnet.net>;tag=00082166efcb002153bc0a31-20004ffd..To: <sip:1001@fs2000.testnet.ne t;user=phone>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-2c94..Call-ID: 00082166-efcb0013-692f44d9-6760a891@10.10.1.144..D ate: Thu, 02 Jun 2011 19:14:14 GMT..CSeq: 101 ACK..Content-Length: 0.... Gary Chen

2 1

discussing about next major release - v3.2.0
by Daniel-Constantin Mierla 02 Jun '11

02 Jun '11

Hello, we should decide the roadmap to next major release v3.2.0, so probably an IRC meeting for some real time chatting would be good. I am thinking of June 3 or 4, 15:00GMT. Anyone interested in the meeting and available in one of those dates? The date with most of devs available will be selected. Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

4 7

RTP traffic
by Lucas Alvarez 01 Jun '11

01 Jun '11

Hi, I'm having a problem with RTP traffic, I have softphone connecting from the outside, I can register but Kamailio is sending all RTP traffic to the private IP address that the softphone has in it own local network. How can I fix this? Any help will be appreciated. Lucas

2 2

serweb & xSER 3.1
by Skyler 01 Jun '11

01 Jun '11

Hi all, Wondering if anyone has serweb working/functioning with Kamailio or SER 3.1 branch? If so which version of serweb are you using 2.0.0 or 2.1.0-dev1? Any issues or tips for first time install/setup of serweb? S.

1 0

has_tran_tmcbs function
by Bruno Bresciani 01 Jun '11

01 Jun '11

Hi, I can't understand the has_tran_tmcbs function... Can someone tell me how it works? Regards,

2 1

Kamailio Tables with index
by Klaus Darilion 01 Jun '11

01 Jun '11

Hi! With Openser 1.2 all the tables got an auto-increment-index column. Is this index column just for layout/performance purposes or is it also used somewhere insides the Kamailio code? If not (what I expect), then Kamailio 3.1 should still work with tables without those columns - correct? Thanks Klaus

2 1

ClueCon 2011 News and Notes
by marketing＠cluecon.com 31 May '11

31 May '11

1 0

Issue using pua_bla module
by Watkins, Bradley 31 May '11

31 May '11

I'm trying to get the pua_bla module to do its thing, and I'm running into an issue. I'm getting these errors when the first NOTIFY comes from the phone: ERROR: presence [publish.c:515]: Missing or unsupported event header field value ERROR: presence [publish.c:518]: event=[dialog;sla] I see the phone register and the subsequent SUBSCRIBE (both successful) from Kamailio, so this bit seems to be correct. I'm loading what I think are all the necessary modules: loadmodule "presence.so" loadmodule "presence_xml.so" loadmodule "pua.so" loadmodule "pua_bla.so" And I'm handling the NOTIFY as per the documentation (xlogs to see that in fact the script executes): if(is_method("NOTIFY") && to_uri=~"bla_aor(a)domain.com") { xlog("Matched bla_aor to_uri\n"); if(bla_handle_notify()) { xlog("Handled NOTIFY\n"); t_reply("200", "OK"); } } I see the log entries from the above xlogs, so it certainly is executing this part of the script. Any ideas why I'm getting the errors? All ideas and/or assistance are greatly appreciated. Thanks in advance, - Brad The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it.

3 6

Siremis: changes to Kamailio DB
by Klaus Darilion 31 May '11

31 May '11

Hi! I just read the Siremis installation howto. One of the task of the installer is: "Update SIP Database - this updates the structure of Kamailio database for accounting and charts modules" I wonder what siremis exactly does to the DB. Is greped over the siremis source file but couldn't find the changes. Are they somewhere documented? (e.g. .sql file with the statemens). Anyway, I wonder changes are needed at all. IMO, if possible, the Kamailio DB should not be changed by an external too - otherwise this may cause problems with updates to the Kamailio DB structure. Thanks Klaus

3 3

kamailio debian package bug report : 3.1.3+lenny2
by Philippe HENSEL 31 May '11

31 May '11

Hi developpers and packagers, The file "/usr/lib64/kamailio/libsrdb2.so.1.0" seems to be included in both "kamailio_3.1.3+lenny2_amd64.deb" and "kamailio-mysql-modules_3.1.3 +lenny2_amd64.deb". So it is not possible to install "kamailio-mysql-modules". Thanks for help

5 8

Multidomains - quick question
by Lucas Alvarez 31 May '11

31 May '11

Hi, if I want to register clients from outside of my local network, do I have to enable the domain module and add the external IP as a new domain? Thanks in advance, Lucas Alvarez

2 2

acc Locks
by Borin 31 May '11

31 May '11

Hi all, Recently I met the following problem with kamailio acc table. We are using kamailio for calls dispatching to a couple of rtp gateways and for call statistics acc is used (with mysql). I tried to delete some old entries from acc table and for this time mysql "show processes" showed a lot of entries like " kamailio | Query | 0 | System lock | insert into acc (method,from_tag,to_tag,callid,sip_code,sip_reason,start_time,src_addr,dst_addr,src_ |". I think that meant that kamailio could not write to the acc table for the time the data were being deleted. And it also did not forward INVITE requests further. Does that mean that if kamailio does not manage to write acc record for any reason it does not forward that INVITE/200ok? Best Regards, Kate

2 4

Green VoIP - energy efficiency and performaces of v3.0
by Daniel-Constantin Mierla 31 May '11

31 May '11

Hello, Jan Janak conducted a very interesting research project regarding energy efficiency of VoIP systems during 2010, a collaboration between iptel.org and Columbia University. The team used the source code from sip-router.org GIT repository from January 2010, which corresponds to Kamailio (former OpenSER) and SER v3.0. The latest stable series v3.1 shares the same internal architecture with v3.0. As part of the research work, Jan could also gather some figures about capacity and performances of v3.0 with a quite complex configuration file: etc/sip-router-oob.cfg (involving authentication and NAT traversal as well). You can read the paper about energy efficiency at: - Green VoIP Article: http://asipto.com/u/2j The draft notes about capacity and performances of v3.0 are available at: - Performances and Capacity for v3.0 Wiki page: http://asipto.com/u/2k Some interesting results: - one instance of SIP server with 500 000 online users (mixed users – behind and not NAT routers) – consumed energy 210W - one instance of SIP server with 1 000 000 online users (no NAT involved) – consumed energy 190W - on a 32-bit machine with 4GB of memory and with 2.5GB reserved for SIP server, the server could support 43 000 simultaneous TLS connections – consumed energy 203W - one SIP server instance with 80 000 permanent TCP connections, the SIP server could still handle at least 1000 requests per second and a connection arrival rate of 1000 new connections per second, done for 20 000 new connections. CPU load generated by the SIP server was from 6% to 8%. I added a new section to the draft notes to list the enhancements done for the latest stable release (v3.1.x) that contribute to performance improvements, like asynchronous TLS, fine tuning of memory for TLS connections and raw UDP sockets. Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

5 9

Late parallel branch (without canceling previous pending branches)
by Iñaki Baz Castillo 31 May '11

31 May '11

Hi, just wondering whether the following scenario (which it's not possible in kamailio 1.5) is possible in 3.X: 1) Alice calls a PSTN number. 2) Proxy receives the call and routes it to a media server which plays an early-announcement (30 seconds long). 3) After N seconds (maybe 10 seconds) proxy starts a new parallel branch to the PSTN gateway *without* cancelling the previous branch with the media server. 4) Let's suppose the PSTN phone rings and proxy gets 183 with SDP (ringing in real audio). Proxy converts such 183 into a 180 with no SDP (already possible in Kamailio). 5) After 30 seconds media server rejects the call with 480. PSTN branch remains ringing as usual. I just would like to know if Kamailio 3.X allows point 3. I think it's hard but maybe I miss some feature. Thanks a lot. -- Iñaki Baz Castillo <ibc(a)aliax.net>

2 2

Dispatcher variables
by Lee Archer 31 May '11

31 May '11

Hi all, I am using dispatcher and on a failure I would like to log the server that has failed. I am using $dd but this only ever returns the address of the first server that was tried and not any subsequent servers that were tried and also failed. What should I be using to display the IP address of the server dispatcher is about to use? Thanks Lee

2 2

Fw: Load capacity of SER??
by Piyush.Bansal＠relianceada.com 31 May '11

31 May '11

Dear All, I'm using SER as a SIP proxy server for quite a long time. I'm not been able to find its load bearing capacity. If you can help me get a figure in terms of BHCA, that'll be quite helpful. Thanks and Regards, --Piyush Bansal The information contained in this e-mail message is intended only for the use of the individual or entity to which it is addressed. If you are not the intended recipient, you should return it to the sender immediately. Please note that while we scan all e-mails for viruses we cannot guarantee that any e-mail is virus-free and accept no liability for any damage caused by any virus transmitted by this email.

1 0

IMC module doesn't save data to mysql database
by Ozren Lapcevic 30 May '11

30 May '11

Hi all, I'm trying to configure IMC module and get it to work properly but I've run into problems with saving IMC module data to mysql database. I'm using following code provided in IMC module documentation: modparam("imc", "db_url", DBURL) ## route block ### if(is_method("MESSAGE") && (uri=~ "sip:chat-[0-9]+@") || (uri=~ "sip:chat-manager@")) { xlog("L_INFO", "going to imc: F [$fu] T [$tu] B [$rb]\n"); if(imc_manager()) sl_send_reply("200", "ok - imc_manager()"); else sl_send_reply("500", "command error"); exit; } I am creating chat room with following message sent to chat-manager(a)abc.hr: "#create chat-111" DEBUG log returns following: May 30 18:01:23 ser-test /usr/local/kamailio-ozren/sbin/kamailio[13225]: INFO: <script>: going to imc: F [sip:oz@abc.hr] T [sip:chat-manager@abc.hr] B [#create chat-111] May 30 18:01:23 ser-test /usr/local/kamailio-ozren/sbin/kamailio[13225]: DEBUG: imc [imc.c:525]: found command May 30 18:01:23 ser-test /usr/local/kamailio-ozren/sbin/kamailio[13225]: DEBUG: imc [imc_cmd.c:146]: command: [create] May 30 18:01:23 ser-test /usr/local/kamailio-ozren/sbin/kamailio[13225]: DEBUG: imc [imc_cmd.c:151]: parameter 0=[chat-111] May 30 18:01:23 ser-test /usr/local/kamailio-ozren/sbin/kamailio[13225]: DEBUG: imc [imc_cmd.c:171]: new room [chat-111] May 30 18:01:23 ser-test /usr/local/kamailio-ozren/sbin/kamailio[13225]: DEBUG: imc [imc_cmd.c:186]: added room uri= sip:chat-111@abc.hr May 30 18:01:23 ser-test /usr/local/kamailio-ozren/sbin/kamailio[13225]: DEBUG: imc [imc_mng.c:322]: [uri]= sip:oz@abc.hr May 30 18:01:23 ser-test /usr/local/kamailio-ozren/sbin/kamailio[13225]: DEBUG: imc [imc_mng.c:326]: [user]= oz May 30 18:01:23 ser-test /usr/local/kamailio-ozren/sbin/kamailio[13225]: DEBUG: imc [imc_cmd.c:196]: added the owner as the first member [ sip:oz@abc.hr] "./kamctl fifo imc_list_rooms" returns following: ROOM:: URI=sip:chat-111@abc.hr MEMBERS=0 OWNER=sip:oz@abc.hr "./kamctl fifo imc_list_members sip:chat-111@abc.hr" returns following: ROOM:: sip:chat-111@abc.hr NR_OF_MEMBERS=1 MEMBER:: sip:oz@abc.hr However, that data is not persisted/saved in the imc_rooms and imc_members tables in the database. Any clues what might be the problem here? Any help is greatly appreciated. Best regards, Ozren Lapcevic PS I'm runing Kamailio 3.1.4 with default cfg file with enabled mysql, multidomain, usrlocdb, auth and NAT. I can register clients, establish calls, send IMs, etc. imc_rooms and imc_members tables were created when kamailio was installed. I've also checked imc-create.sql script and tables are correctly created. The same DBURL used for imc is also used for auth_db, domain and usrloc modules and I haven't experienced any problems with those modules and database connection/persistence. domain, location, version and subscriber tables are correctly populated. I'm not getting any errors from imc_manager(), it returns 200 OK. I can't trace any errors in DEBUG related to problems with mysql database.

1 0

Kamailio service not starting on reboot when using mysql database
by Rick Aldred 30 May '11

30 May '11

I have set up kamailio 3.1.2 on an install of Debian Squeeze, using the default config but with the following defines enabled: WITH_MYSQLWITH_AUTHWITH_USRLOCDBWITH_NAT Everything works as expected when starting Kamailio from a terminal, e.g.service kamailio start But after a reboot, the service fails to start and all SIP requests are dropped with a ICMP port unreachable error. The following is an extract of /var/log/syslog when this problem occurs. It looks like Kamailio is failing to access the mysql socket, as the mysql service has not yet started? Apr 12 10:08:31 SIPServer /usr/sbin/kamailio[1347]: ERROR: db_mysql [km_my_con.c:109]: driver error: Can't connect to local MySQL server through socket '/var/run/mysqld/mysqld.sock' (2)Apr 12 10:08:31 SIPServer /usr/sbin/kamailio[1347]: ERROR: <core> [db.c:289]: could not add connection to the poolApr 12 10:08:31 SIPServer /usr/sbin/kamailio[1347]: ERROR: usrloc [dlist.c:486]: failed to open database connectionApr 12 10:08:31 SIPServer /usr/sbin/kamailio[1347]: ERROR: registrar [reg_mod.c:396]: failed to register domainApr 12 10:08:31 SIPServer /usr/sbin/kamailio[1347]: ERROR: <core> [route.c:1161]: fixing failed (code=-1) at cfg:/etc/kamailio/kamailio.cfg:569Apr 12 10:08:31 SIPServer /usr/sbin/kamailio[1347]: ERROR: <core> [route.c:1161]: fixing failed (code=-1) at cfg:/etc/kamailio/kamailio.cfg:572Apr 12 10:08:31 SIPServer /usr/sbin/kamailio[1347]: ERROR: <core> [route.c:1161]: fixing failed (code=-1) at cfg:/etc/kamailio/kamailio.cfg:574Apr 12 10:08:34 SIPServer mysqld_safe: Starting mysqld daemon with databases from /var/lib/mysqlApr 12 10:08:34 SIPServer mysqld: 110412 10:08:34 [Note] Plugin 'FEDERATED' is disabled.Apr 12 10:08:36 SIPServer mysqld: 110412 10:08:36 InnoDB: Started; log sequence number 0 44233Apr 12 10:08:36 SIPServer mysqld: 110412 10:08:36 [Note] Event Scheduler: Loaded 0 eventsApr 12 10:08:36 SIPServer mysqld: 110412 10:08:36 [Note] /usr/sbin/mysqld: ready for connections.Apr 12 10:08:36 SIPServer mysqld: Version: '5.1.49-3' socket: '/var/run/mysqld/mysqld.sock' port: 3306 (Debian) Both the Kamailio and mysql services start with priority 18 on run levels 2345. I tried adjusting the Kamailio service to priority 19, but the problem still occurs. I am relatively new to Kamailio and mysql as well as Linux / Debian administration, so I'm not really sure how to solve this problem. Any ideas would be greatly appreciated. Regards,Rick Aldred

5 10

kamailio and radius
by Asun 30 May '11

30 May '11

Hello I've installed kamailio 3.1 but It's doesn't Work with radius authentication. I've followed this manual <http://www.kamailio.org/docs/openser-radius-1.0.x.html> http://www.kamailio.org/docs/openser-radius-1.0.x.html but it does'nt work, because is for openser not kamailio. Is there another manual to do this? Anyone have radius authentication with kamailio? Can anyone help me? Thank you Asun

3 8

Presence changes not seen for first time login
by Swetha Raj 30 May '11

30 May '11

Hi All, I have Kamailio3.1 set up by following the steps mentioned here - http://kb.asipto.com/kamailio:presence:k31-made-simple. When i login two users via Jitsi and add each other,one user is not able to see the other user's presence changes.Below are the steps to reproduce - 1.Login user1@sipdomain via jitsi 2.Login user2@sipdomain vis jitsi 3.user1 adds user2,user2 receives the request and accepts it. 4.user2 adds user1. 5.Now any presence changes from user1 is visible to user2 But any changes from user2 is not visible to user1. This issue is resolved if the users re-login.Any ideas what could be the issue ? Also how do i see the complete SIP headers in kamailio log,is their a way to enable it ? Thanks, Swetha

3 4

Invitation to connect on LinkedIn
by Anand Kumar via LinkedIn 28 May '11

28 May '11

LinkedIn ------------ Anand Kumar requested to add you as a connection on LinkedIn: ------------------------------------------ Luis, I'd like to add you to my professional network on LinkedIn. - Anand Accept invitation from Anand Kumar http://www.linkedin.com/e/-y8k3ym-go88enz1-1n/XkppUi8mK5TI9VJsZIpnfBovln4Su… View invitation from Anand Kumar http://www.linkedin.com/e/-y8k3ym-go88enz1-1n/XkppUi8mK5TI9VJsZIpnfBovln4Su… ------------------------------------------ DID YOU KNOW you can conduct a more credible and powerful reference check using LinkedIn? Enter the company name and years of employment or the prospective employee to find their colleagues that are also in your network. This provides you with a more balanced set of feedback to evaluate that new hire. http://www.linkedin.com/e/-y8k3ym-go88enz1-1n/rsr/inv-27/ -- (c) 2011, LinkedIn Corporation

1 0

core kamailio 3.1.2
by Bruno Bresciani 27 May '11

27 May '11

Hi ALL Why kamailio 3.1.2 generate core when function recvmsg return 0? Below is the trace of the core kamailio: May 27 10:29:14 sswpst00 /home2/local/kamailio/sbin/kamailio[31288]: INFO: <script>: CFG: Cseq INVITE ... May 27 10:29:14 sswpst00 /home2/local/kamailio/sbin/kamailio[31288]: DEBUG: tm [t_reply.c:1129]: ->>>>>>>>> T_code=100, new_code=488 May 27 10:29:14 sswpst00 /home2/local/kamailio/sbin/kamailio[31288]: DEBUG: tm [t_hooks.c:288]: DBG: trans=0xb61304a0, callback type 64, id 0 entered *May 27 10:29:15 sswpst00 /home2/local/kamailio/sbin/kamailio[31308]: : <core> [pass_fd.c:293]: ERROR: receive_fd: EOF on 11* May 27 10:29:15 sswpst00 /home2/local/kamailio/sbin/kamailio[31308]: DEBUG: <core> [tcp_main.c:3551]: DBG: handle_ser_child: dead child 1, pid 31288 (shutting down?) May 27 10:29:15 sswpst00 /home2/local/kamailio/sbin/kamailio[31308]: DEBUG: <core> [io_wait.h:617]: DBG: io_watch_del (0x82561e0, 11, -1, 0x0) fd_no=18 called *May 27 10:29:15 sswpst00 /home2/local/kamailio/sbin/kamailio[31285]: ALERT: <core> [main.c:741]: child process 31288 exited by a signal 11* May 27 10:29:15 sswpst00 /home2/local/kamailio/sbin/kamailio[31285]: ALERT: <core> [main.c:744]: core was generated May 27 10:29:15 sswpst00 /home2/local/kamailio/sbin/kamailio[31285]: INFO: <core> [main.c:756]: INFO: terminating due to SIGCHLD *May 27 10:29:15 sswpst00 /home2/local/kamailio/sbin/kamailio[31308]: INFO: <core> [main.c:807]: INFO: signal 15 received* May 27 10:29:15 sswpst00 /home2/local/kamailio/sbin/kamailio[31308]: DEBUG: <core> [main.c:818]: Memory status (pkg): Regards,

1 0

ser_ctl / serweb
by Claudio Furrer 27 May '11

27 May '11

Hi all, Just want to ask if ser_ctl application will be merged into sr3.1 or future releases. By the moment I've found it here [1], [2] or [3]. Is it full compatible with sip-router v3.x? I know kamctl and kamdbctl would be suggested but these are for kamailio flavour. I don't find too much work based on SER flavour regarding to database/users administration and web provisioning. BTW, siremis only works with kamailio db structures, but for sip-router v3 ser-flavoured only found serweb 2.x here [4] and [5] which I don't know if is it working well with the current new releases of the project. I would appreciate if someone can advice me if kamailio is the way to go, because have find that sip-router or ser flavours miss some docs and apps that kamailio already have. I'm asking from a a SER-user point of view, who want to migrate to SR-3 :( [1] http://ftp.iptel.org/pub/serctl/daily-snapshots/ [2] http://git.sip-router.org/cgi-bin/gitweb.cgi?p=ser;a=tree;f=ser_ctl;h=7db4a… [3] http://cvs.berlios.de/viewvc/ser/serctl/ [4] http://ftp.iptel.org/pub/serweb/daily-snapshots/ [5] http://developer.berlios.de/projects/serweb/ Thank you, Claudio

3 2

Howto restore a FIFO file?
by Spinov Evgeniy 27 May '11

27 May '11

Hello, everyone. I've accidentally started 2nd copy of Kamailio and killed it at once. However, fifo file was deleted by killed Kamailio, while the first one copy is running fine. Is there any way to restore the fifo file to the first kamailio, unless restarting it? Thank you.

3 2

[OT] IETF SIMPLE WG will destroy MSRP with the new draft-ietf-simple-msrp-sessmatch-11
by Iñaki Baz Castillo 27 May '11

27 May '11

Hi, for those interested in MSRP protocol (instant message sessions and file transfer for SIP) there are bad news: Even if there are already clients and servers (MSRP relays o IM conference servers) implementing the MSRP protocol (RFC 4975 and 4976) the SIMPLE WG will publish a new draft [*] that breaks these RFC's just to satisfy big vendors interested in deploying MSRP capable SBC/ALG boxes (so instead of solving NAT issues with MSRP relays as RFC 4976 states, they want it to be fixed in the router by doing ugly ALG, or in a SBC). This is terrible because all the MSRP devices should implement this new draft in order to interoperate (no backward compatibility at all). The draft also breaks the security defined in RFC 4975 (for example, TLS name based authentication cannot work anymore). For further information I recommend reading these two posts: http://blog.tekelec.com/blog/bid/29816/More-on-MSRP-Session-Match-Extension http://blog.tekelec.com/blog/bid/33138/MSRP-Session-Match-Backwards-Compati… and also these very *hot* mail threads in the SIMPLE maillist: http://www.ietf.org/mail-archive/web/simple/current/msg09227.html http://www.ietf.org/mail-archive/web/simple/current/msg09229.html [*] http://tools.ietf.org/html/draft-ietf-simple-msrp-sessmatch-11 -- Iñaki Baz Castillo <ibc(a)aliax.net>

3 23

Re: [SR-Users] sr-users Digest, Vol 72, Issue 67
by Boris 27 May '11

27 May '11

sr-users-request(a)lists.sip-router.org wrote: >Send sr-users mailing list submissions to > sr-users(a)lists.sip-router.org > >To subscribe or unsubscribe via the World Wide Web, visit > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.sip-router.org > >You can reach the person managing the list at > sr-users-owner(a)lists.sip-router.org > >When replying, please edit your Subject line so it is more specific >than "Re: Contents of sr-users digest..." > > >Today's Topics: > > 1. Aliases Using the Default Script; (David J.) > 2. Re: Green VoIP - energy efficiency and performances of v3.0 > (Jan Janak) > 3. Kamailio v3.1.4 Released (Daniel-Constantin Mierla) > 4. Re: ser_ctl / serweb (caio) > 5. Re: [OT] IETF SIMPLE WG will destroy MSRP with the new > draft-ietf-simple-msrp-sessmatch-11 (I?aki Baz Castillo) > 6. Re: [OT] IETF SIMPLE WG will destroy MSRP with the new > draft-ietf-simple-msrp-sessmatch-11 (Daniel-Constantin Mierla) > > >---------------------------------------------------------------------- > >Message: 1 >Date: Thu, 26 May 2011 14:27:55 -0400 >From: "David J." <david(a)styleflare.com> >Subject: [SR-Users] Aliases Using the Default Script; >To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List" <sr-users(a)lists.sip-router.org> >Message-ID: <4DDE9BAB.4060401(a)styleflare.com> >Content-Type: text/plain; charset=ISO-8859-1; format=flowed > >I am using the kamailio default script on 3.1.3. > >I was wondering what happens when I added an Alias in dbaliases? > >For example if I add 18005551212(a)mydomain.com alias to 1001(a)mydomain.com > >when an invite comes in; it works perfect I got a 200 back. (1001 Device >rings.) > >If I add another alias to a remote server; ie. > >18005551213(a)mydomain.com alias to 1800555124(a)someotherdomain.com > >I get a 404 back. > >I wonder what happens internally? > >What should I modify to enable this case; > >Thanks. > > > >------------------------------ > >Message: 2 >Date: Thu, 26 May 2011 20:49:39 +0200 >From: Jan Janak <jan(a)ryngle.com> >Subject: Re: [SR-Users] Green VoIP - energy efficiency and > performances of v3.0 >To: miconda(a)gmail.com >Cc: "SIP Router - Kamailio $OpenSER$ and SIP Express Router $SER$ > - Users Mailing List" <sr-users(a)lists.sip-router.org>, sr-dev > <sr-dev(a)lists.sip-router.org> >Message-ID: <BANLkTikUb_iwBDEuqk977-ECmmHo4dEXvQ(a)mail.gmail.com> >Content-Type: text/plain; charset=UTF-8 > >No, I turned it off. > >-Jan > >On Thu, May 26, 2011 at 11:50, Daniel-Constantin Mierla ><miconda(a)gmail.com> wrote: >> Hello, >> >> out of curiosity, since you used the sources from GIT - was memory debugging >> on? It is usually enabled in master branch and that could have some impact >> in memory usage and performances... >> >> Thanks, >> Daniel >> >> On 5/25/11 3:00 PM, Jan Janak wrote: >>> >>> On Wed, May 25, 2011 at 06:54, Jeremya<jeremy(a)electrosilk.net> ?wrote: >>>> >>>> These figures pale into insignificance compared to the power required >>>> for standard SIP devices - typically 5-8 watts per device multiplied by >>>> the number of devices. >>>> >>>> When you factor in Gigabit Ethernet the power ups significantly. >>>> >>>> Optimisation at the server level is not significant on any scale. >>>> Optimisation on communications power: i.e. end-devices, DSL& ?switches >>>> is where the power savings are important. >>> >>> Sure, the total power consumption of the whole system is dominated by >>> the power consumption of end-point devices, there's no doubt about >>> that and the paper says that. >>> >>> Nevertheless, as an ITSP you are typically paying for the energy >>> consumed by your servers and in that case knowing what you can expect >>> and how many servers you need is useful. Modern data-center servers >>> have significant base-line power consumption and a portion of that >>> needs to be attributed to the SIP service running on those servers. >>> >>> -Jan >>> >>>> Daniel-Constantin Mierla wrote: >>>>> >>>>> Hello, >>>>> >>>>> Jan Janak conducted a very interesting research project regarding >>>>> energy efficiency of VoIP systems during 2010, a collaboration between >>>>> iptel.org and Columbia University. >>>>> >>>>> The team used the source code from sip-router.org GIT repository from >>>>> January 2010, which corresponds to Kamailio (former OpenSER) and SER >>>>> v3.0. The latest stable series v3.1 shares the same internal >>>>> architecture with v3.0. >>>>> >>>>> As part of the research work, Jan could also gather some figures about >>>>> capacity and performances of v3.0 with a quite complex configuration >>>>> file: etc/sip-router-oob.cfg (involving authentication and NAT >>>>> traversal as well). >>>>> >>>>> You can read the paper about energy efficiency at: >>>>> >>>>> - Green VoIP Article: http://asipto.com/u/2j >>>>> >>>>> The draft notes about capacity and performances of v3.0 are available >>>>> at: >>>>> >>>>> - Performances and Capacity for v3.0 Wiki page: http://asipto.com/u/2k >>>>> >>>>> Some interesting results: >>>>> >>>>> - one instance of SIP server with 500 000 online users (mixed users ? >>>>> behind and not NAT routers) ? consumed energy 210W >>>>> - one instance of SIP server with 1 000 000 online users (no NAT >>>>> involved) ? consumed energy 190W >>>>> - on a 32-bit machine with 4GB of memory and with 2.5GB reserved for >>>>> SIP server, the server could support 43 000 simultaneous TLS >>>>> connections ? consumed energy 203W >>>>> - one SIP server instance with 80 000 permanent TCP connections, the >>>>> SIP server could still handle at least 1000 requests per second and a >>>>> connection arrival rate of 1000 new connections per second, done for >>>>> 20 000 new connections. CPU load generated by the SIP server was from >>>>> 6% to 8%. >>>>> >>>>> I added a new section to the draft notes to list the enhancements done >>>>> for the latest stable release (v3.1.x) that contribute to performance >>>>> improvements, like asynchronous TLS, fine tuning of memory for TLS >>>>> connections and raw UDP sockets. >>>>> >>>>> Cheers, >>>>> Daniel >>>>> >>>> >>>> _______________________________________________ >>>> 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 >> >> -- >> Daniel-Constantin Mierla >> http://www.asipto.com >> >> > > > >------------------------------ > >Message: 3 >Date: Thu, 26 May 2011 21:00:42 +0200 >From: Daniel-Constantin Mierla <miconda(a)gmail.com> >Subject: [SR-Users] Kamailio v3.1.4 Released >To: kamailio <sr-users(a)lists.sip-router.org>, sr-dev > <sr-dev(a)lists.sip-router.org>, business(a)lists.kamailio.org >Message-ID: <4DDEA35A.7070606(a)gmail.com> >Content-Type: text/plain; charset=ISO-8859-1; format=flowed > >Hello, > >Kamailio SIP Server v3.1.4 stable release is out. > >This is a maintenance release of latest stable branch, 3.1, that >includes fixes since release of v3.1.3. There is no change to database >schema or configuration language structure. Deployments running previous >v3.1.x versions are strongly recommended to be upgraded to v3.1.4. > >For more details about version 3.1.4, visit: > >http://www.kamailio.org/w/2011/05/kamailio-v3-1-4-released/ > >Cheers, >Daniel > >-- >Daniel-Constantin Mierla -- http://www.asipto.com >http://linkedin.com/in/miconda -- http://twitter.com/miconda > > > > >------------------------------ > >Message: 4 >Date: Thu, 26 May 2011 17:59:42 -0300 >From: caio <elcaio(a)gmail.com> >Subject: Re: [SR-Users] ser_ctl / serweb >To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List" <sr-users(a)lists.sip-router.org> >Message-ID: <BANLkTik+dnTp-egK2xy1ua=tPnCcaDuuOA(a)mail.gmail.com> >Content-Type: text/plain; charset="utf-8" > >Hello, > >Anyone know if ser_ctl (python utility) will be merged into master? Or is >going to be deprecated? >If it still alive, are there any examples/docs of the usage for DB users >provisoining? >kamctl and kamdbctl are recomended for its substitution? > >Which utility will be official for sip-router? >Sorry if it already were discussed, but I didn't find comments about it >since 2010. > >Thank you >Claudio > >On Tue, May 24, 2011 at 11:14 AM, Claudio Furrer <elcaio(a)gmail.com> wrote: > >> Hi all, >> >> Just want to ask if ser_ctl application will be merged into sr3.1 or future >> releases. >> By the moment I've found it here [1], [2] or [3]. >> >> Is it full compatible with sip-router v3.x? >> >> I know kamctl and kamdbctl would be suggested but these are for kamailio >> flavour. I don't find too much work based on SER flavour regarding to >> database/users administration and web provisioning. >> BTW, siremis only works with kamailio db structures, but for sip-router v3 >> ser-flavoured only found serweb 2.x here [4] and [5] which I don't know if >> is >> it working well with the current new releases of the project. >> >> I would appreciate if someone can advice me if kamailio is the way to go, >> because have find that sip-router or ser flavours miss some docs and apps >> that >> kamailio already have. >> >> I'm asking from a a SER-user point of view, who want to migrate to SR-3 :( >> >> [1] http://ftp.iptel.org/pub/serctl/daily-snapshots/ >> [2] >> >> http://git.sip-router.org/cgi-bin/gitweb.cgi?p=ser;a=tree;f=ser_ctl;h=7db4a… >> [3] http://cvs.berlios.de/viewvc/ser/serctl/ >> >> [4] http://ftp.iptel.org/pub/serweb/daily-snapshots/ >> [5] http://developer.berlios.de/projects/serweb/ >> >> Thank you, >> Claudio >> >

1 0

Re: [SR-Users] sr-users Digest, Vol 72, Issue 67
by Boris 27 May '11

27 May '11

sr-users-request(a)lists.sip-router.org wrote: >Send sr-users mailing list submissions to > sr-users(a)lists.sip-router.org > >To subscribe or unsubscribe via the World Wide Web, visit > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.sip-router.org > >You can reach the person managing the list at > sr-users-owner(a)lists.sip-router.org > >When replying, please edit your Subject line so it is more specific >than "Re: Contents of sr-users digest..." > > >Today's Topics: > > 1. Aliases Using the Default Script; (David J.) > 2. Re: Green VoIP - energy efficiency and performances of v3.0 > (Jan Janak) > 3. Kamailio v3.1.4 Released (Daniel-Constantin Mierla) > 4. Re: ser_ctl / serweb (caio) > 5. Re: [OT] IETF SIMPLE WG will destroy MSRP with the new > draft-ietf-simple-msrp-sessmatch-11 (I?aki Baz Castillo) > 6. Re: [OT] IETF SIMPLE WG will destroy MSRP with the new > draft-ietf-simple-msrp-sessmatch-11 (Daniel-Constantin Mierla) > > >---------------------------------------------------------------------- > >Message: 1 >Date: Thu, 26 May 2011 14:27:55 -0400 >From: "David J." <david(a)styleflare.com> >Subject: [SR-Users] Aliases Using the Default Script; >To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List" <sr-users(a)lists.sip-router.org> >Message-ID: <4DDE9BAB.4060401(a)styleflare.com> >Content-Type: text/plain; charset=ISO-8859-1; format=flowed > >I am using the kamailio default script on 3.1.3. > >I was wondering what happens when I added an Alias in dbaliases? > >For example if I add 18005551212(a)mydomain.com alias to 1001(a)mydomain.com > >when an invite comes in; it works perfect I got a 200 back. (1001 Device >rings.) > >If I add another alias to a remote server; ie. > >18005551213(a)mydomain.com alias to 1800555124(a)someotherdomain.com > >I get a 404 back. > >I wonder what happens internally? > >What should I modify to enable this case; > >Thanks. > > > >------------------------------ > >Message: 2 >Date: Thu, 26 May 2011 20:49:39 +0200 >From: Jan Janak <jan(a)ryngle.com> >Subject: Re: [SR-Users] Green VoIP - energy efficiency and > performances of v3.0 >To: miconda(a)gmail.com >Cc: "SIP Router - Kamailio $OpenSER$ and SIP Express Router $SER$ > - Users Mailing List" <sr-users(a)lists.sip-router.org>, sr-dev > <sr-dev(a)lists.sip-router.org> >Message-ID: <BANLkTikUb_iwBDEuqk977-ECmmHo4dEXvQ(a)mail.gmail.com> >Content-Type: text/plain; charset=UTF-8 > >No, I turned it off. > >-Jan > >On Thu, May 26, 2011 at 11:50, Daniel-Constantin Mierla ><miconda(a)gmail.com> wrote: >> Hello, >> >> out of curiosity, since you used the sources from GIT - was memory debugging >> on? It is usually enabled in master branch and that could have some impact >> in memory usage and performances... >> >> Thanks, >> Daniel >> >> On 5/25/11 3:00 PM, Jan Janak wrote: >>> >>> On Wed, May 25, 2011 at 06:54, Jeremya<jeremy(a)electrosilk.net> ?wrote: >>>> >>>> These figures pale into insignificance compared to the power required >>>> for standard SIP devices - typically 5-8 watts per device multiplied by >>>> the number of devices. >>>> >>>> When you factor in Gigabit Ethernet the power ups significantly. >>>> >>>> Optimisation at the server level is not significant on any scale. >>>> Optimisation on communications power: i.e. end-devices, DSL& ?switches >>>> is where the power savings are important. >>> >>> Sure, the total power consumption of the whole system is dominated by >>> the power consumption of end-point devices, there's no doubt about >>> that and the paper says that. >>> >>> Nevertheless, as an ITSP you are typically paying for the energy >>> consumed by your servers and in that case knowing what you can expect >>> and how many servers you need is useful. Modern data-center servers >>> have significant base-line power consumption and a portion of that >>> needs to be attributed to the SIP service running on those servers. >>> >>> -Jan >>> >>>> Daniel-Constantin Mierla wrote: >>>>> >>>>> Hello, >>>>> >>>>> Jan Janak conducted a very interesting research project regarding >>>>> energy efficiency of VoIP systems during 2010, a collaboration between >>>>> iptel.org and Columbia University. >>>>> >>>>> The team used the source code from sip-router.org GIT repository from >>>>> January 2010, which corresponds to Kamailio (former OpenSER) and SER >>>>> v3.0. The latest stable series v3.1 shares the same internal >>>>> architecture with v3.0. >>>>> >>>>> As part of the research work, Jan could also gather some figures about >>>>> capacity and performances of v3.0 with a quite complex configuration >>>>> file: etc/sip-router-oob.cfg (involving authentication and NAT >>>>> traversal as well). >>>>> >>>>> You can read the paper about energy efficiency at: >>>>> >>>>> - Green VoIP Article: http://asipto.com/u/2j >>>>> >>>>> The draft notes about capacity and performances of v3.0 are available >>>>> at: >>>>> >>>>> - Performances and Capacity for v3.0 Wiki page: http://asipto.com/u/2k >>>>> >>>>> Some interesting results: >>>>> >>>>> - one instance of SIP server with 500 000 online users (mixed users ? >>>>> behind and not NAT routers) ? consumed energy 210W >>>>> - one instance of SIP server with 1 000 000 online users (no NAT >>>>> involved) ? consumed energy 190W >>>>> - on a 32-bit machine with 4GB of memory and with 2.5GB reserved for >>>>> SIP server, the server could support 43 000 simultaneous TLS >>>>> connections ? consumed energy 203W >>>>> - one SIP server instance with 80 000 permanent TCP connections, the >>>>> SIP server could still handle at least 1000 requests per second and a >>>>> connection arrival rate of 1000 new connections per second, done for >>>>> 20 000 new connections. CPU load generated by the SIP server was from >>>>> 6% to 8%. >>>>> >>>>> I added a new section to the draft notes to list the enhancements done >>>>> for the latest stable release (v3.1.x) that contribute to performance >>>>> improvements, like asynchronous TLS, fine tuning of memory for TLS >>>>> connections and raw UDP sockets. >>>>> >>>>> Cheers, >>>>> Daniel >>>>> >>>> >>>> _______________________________________________ >>>> 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 >> >> -- >> Daniel-Constantin Mierla >> http://www.asipto.com >> >> > > > >------------------------------ > >Message: 3 >Date: Thu, 26 May 2011 21:00:42 +0200 >From: Daniel-Constantin Mierla <miconda(a)gmail.com> >Subject: [SR-Users] Kamailio v3.1.4 Released >To: kamailio <sr-users(a)lists.sip-router.org>, sr-dev > <sr-dev(a)lists.sip-router.org>, business(a)lists.kamailio.org >Message-ID: <4DDEA35A.7070606(a)gmail.com> >Content-Type: text/plain; charset=ISO-8859-1; format=flowed > >Hello, > >Kamailio SIP Server v3.1.4 stable release is out. > >This is a maintenance release of latest stable branch, 3.1, that >includes fixes since release of v3.1.3. There is no change to database >schema or configuration language structure. Deployments running previous >v3.1.x versions are strongly recommended to be upgraded to v3.1.4. > >For more details about version 3.1.4, visit: > >http://www.kamailio.org/w/2011/05/kamailio-v3-1-4-released/ > >Cheers, >Daniel > >-- >Daniel-Constantin Mierla -- http://www.asipto.com >http://linkedin.com/in/miconda -- http://twitter.com/miconda > > > > >------------------------------ > >Message: 4 >Date: Thu, 26 May 2011 17:59:42 -0300 >From: caio <elcaio(a)gmail.com> >Subject: Re: [SR-Users] ser_ctl / serweb >To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List" <sr-users(a)lists.sip-router.org> >Message-ID: <BANLkTik+dnTp-egK2xy1ua=tPnCcaDuuOA(a)mail.gmail.com> >Content-Type: text/plain; charset="utf-8" > >Hello, > >Anyone know if ser_ctl (python utility) will be merged into master? Or is >going to be deprecated? >If it still alive, are there any examples/docs of the usage for DB users >provisoining? >kamctl and kamdbctl are recomended for its substitution? > >Which utility will be official for sip-router? >Sorry if it already were discussed, but I didn't find comments about it >since 2010. > >Thank you >Claudio > >On Tue, May 24, 2011 at 11:14 AM, Claudio Furrer <elcaio(a)gmail.com> wrote: > >> Hi all, >> >> Just want to ask if ser_ctl application will be merged into sr3.1 or future >> releases. >> By the moment I've found it here [1], [2] or [3]. >> >> Is it full compatible with sip-router v3.x? >> >> I know kamctl and kamdbctl would be suggested but these are for kamailio >> flavour. I don't find too much work based on SER flavour regarding to >> database/users administration and web provisioning. >> BTW, siremis only works with kamailio db structures, but for sip-router v3 >> ser-flavoured only found serweb 2.x here [4] and [5] which I don't know if >> is >> it working well with the current new releases of the project. >> >> I would appreciate if someone can advice me if kamailio is the way to go, >> because have find that sip-router or ser flavours miss some docs and apps >> that >> kamailio already have. >> >> I'm asking from a a SER-user point of view, who want to migrate to SR-3 :( >> >> [1] http://ftp.iptel.org/pub/serctl/daily-snapshots/ >> [2] >> >> http://git.sip-router.org/cgi-bin/gitweb.cgi?p=ser;a=tree;f=ser_ctl;h=7db4a… >> [3] http://cvs.berlios.de/viewvc/ser/serctl/ >> >> [4] http://ftp.iptel.org/pub/serweb/daily-snapshots/ >> [5] http://developer.berlios.de/projects/serweb/ >> >> Thank you, >> Claudio >> >

1 0

Re: [SR-Users] sr-users Digest, Vol 72, Issue 67
by Boris 27 May '11

27 May '11

sr-users-request(a)lists.sip-router.org wrote: >Send sr-users mailing list submissions to > sr-users(a)lists.sip-router.org > >To subscribe or unsubscribe via the World Wide Web, visit > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.sip-router.org > >You can reach the person managing the list at > sr-users-owner(a)lists.sip-router.org > >When replying, please edit your Subject line so it is more specific >than "Re: Contents of sr-users digest..." > > >Today's Topics: > > 1. Aliases Using the Default Script; (David J.) > 2. Re: Green VoIP - energy efficiency and performances of v3.0 > (Jan Janak) > 3. Kamailio v3.1.4 Released (Daniel-Constantin Mierla) > 4. Re: ser_ctl / serweb (caio) > 5. Re: [OT] IETF SIMPLE WG will destroy MSRP with the new > draft-ietf-simple-msrp-sessmatch-11 (I?aki Baz Castillo) > 6. Re: [OT] IETF SIMPLE WG will destroy MSRP with the new > draft-ietf-simple-msrp-sessmatch-11 (Daniel-Constantin Mierla) > > >---------------------------------------------------------------------- > >Message: 1 >Date: Thu, 26 May 2011 14:27:55 -0400 >From: "David J." <david(a)styleflare.com> >Subject: [SR-Users] Aliases Using the Default Script; >To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List" <sr-users(a)lists.sip-router.org> >Message-ID: <4DDE9BAB.4060401(a)styleflare.com> >Content-Type: text/plain; charset=ISO-8859-1; format=flowed > >I am using the kamailio default script on 3.1.3. > >I was wondering what happens when I added an Alias in dbaliases? > >For example if I add 18005551212(a)mydomain.com alias to 1001(a)mydomain.com > >when an invite comes in; it works perfect I got a 200 back. (1001 Device >rings.) > >If I add another alias to a remote server; ie. > >18005551213(a)mydomain.com alias to 1800555124(a)someotherdomain.com > >I get a 404 back. > >I wonder what happens internally? > >What should I modify to enable this case; > >Thanks. > > > >------------------------------ > >Message: 2 >Date: Thu, 26 May 2011 20:49:39 +0200 >From: Jan Janak <jan(a)ryngle.com> >Subject: Re: [SR-Users] Green VoIP - energy efficiency and > performances of v3.0 >To: miconda(a)gmail.com >Cc: "SIP Router - Kamailio $OpenSER$ and SIP Express Router $SER$ > - Users Mailing List" <sr-users(a)lists.sip-router.org>, sr-dev > <sr-dev(a)lists.sip-router.org> >Message-ID: <BANLkTikUb_iwBDEuqk977-ECmmHo4dEXvQ(a)mail.gmail.com> >Content-Type: text/plain; charset=UTF-8 > >No, I turned it off. > >-Jan > >On Thu, May 26, 2011 at 11:50, Daniel-Constantin Mierla ><miconda(a)gmail.com> wrote: >> Hello, >> >> out of curiosity, since you used the sources from GIT - was memory debugging >> on? It is usually enabled in master branch and that could have some impact >> in memory usage and performances... >> >> Thanks, >> Daniel >> >> On 5/25/11 3:00 PM, Jan Janak wrote: >>> >>> On Wed, May 25, 2011 at 06:54, Jeremya<jeremy(a)electrosilk.net> ?wrote: >>>> >>>> These figures pale into insignificance compared to the power required >>>> for standard SIP devices - typically 5-8 watts per device multiplied by >>>> the number of devices. >>>> >>>> When you factor in Gigabit Ethernet the power ups significantly. >>>> >>>> Optimisation at the server level is not significant on any scale. >>>> Optimisation on communications power: i.e. end-devices, DSL& ?switches >>>> is where the power savings are important. >>> >>> Sure, the total power consumption of the whole system is dominated by >>> the power consumption of end-point devices, there's no doubt about >>> that and the paper says that. >>> >>> Nevertheless, as an ITSP you are typically paying for the energy >>> consumed by your servers and in that case knowing what you can expect >>> and how many servers you need is useful. Modern data-center servers >>> have significant base-line power consumption and a portion of that >>> needs to be attributed to the SIP service running on those servers. >>> >>> -Jan >>> >>>> Daniel-Constantin Mierla wrote: >>>>> >>>>> Hello, >>>>> >>>>> Jan Janak conducted a very interesting research project regarding >>>>> energy efficiency of VoIP systems during 2010, a collaboration between >>>>> iptel.org and Columbia University. >>>>> >>>>> The team used the source code from sip-router.org GIT repository from >>>>> January 2010, which corresponds to Kamailio (former OpenSER) and SER >>>>> v3.0. The latest stable series v3.1 shares the same internal >>>>> architecture with v3.0. >>>>> >>>>> As part of the research work, Jan could also gather some figures about >>>>> capacity and performances of v3.0 with a quite complex configuration >>>>> file: etc/sip-router-oob.cfg (involving authentication and NAT >>>>> traversal as well). >>>>> >>>>> You can read the paper about energy efficiency at: >>>>> >>>>> - Green VoIP Article: http://asipto.com/u/2j >>>>> >>>>> The draft notes about capacity and performances of v3.0 are available >>>>> at: >>>>> >>>>> - Performances and Capacity for v3.0 Wiki page: http://asipto.com/u/2k >>>>> >>>>> Some interesting results: >>>>> >>>>> - one instance of SIP server with 500 000 online users (mixed users ? >>>>> behind and not NAT routers) ? consumed energy 210W >>>>> - one instance of SIP server with 1 000 000 online users (no NAT >>>>> involved) ? consumed energy 190W >>>>> - on a 32-bit machine with 4GB of memory and with 2.5GB reserved for >>>>> SIP server, the server could support 43 000 simultaneous TLS >>>>> connections ? consumed energy 203W >>>>> - one SIP server instance with 80 000 permanent TCP connections, the >>>>> SIP server could still handle at least 1000 requests per second and a >>>>> connection arrival rate of 1000 new connections per second, done for >>>>> 20 000 new connections. CPU load generated by the SIP server was from >>>>> 6% to 8%. >>>>> >>>>> I added a new section to the draft notes to list the enhancements done >>>>> for the latest stable release (v3.1.x) that contribute to performance >>>>> improvements, like asynchronous TLS, fine tuning of memory for TLS >>>>> connections and raw UDP sockets. >>>>> >>>>> Cheers, >>>>> Daniel >>>>> >>>> >>>> _______________________________________________ >>>> 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 >> >> -- >> Daniel-Constantin Mierla >> http://www.asipto.com >> >> > > > >------------------------------ > >Message: 3 >Date: Thu, 26 May 2011 21:00:42 +0200 >From: Daniel-Constantin Mierla <miconda(a)gmail.com> >Subject: [SR-Users] Kamailio v3.1.4 Released >To: kamailio <sr-users(a)lists.sip-router.org>, sr-dev > <sr-dev(a)lists.sip-router.org>, business(a)lists.kamailio.org >Message-ID: <4DDEA35A.7070606(a)gmail.com> >Content-Type: text/plain; charset=ISO-8859-1; format=flowed > >Hello, > >Kamailio SIP Server v3.1.4 stable release is out. > >This is a maintenance release of latest stable branch, 3.1, that >includes fixes since release of v3.1.3. There is no change to database >schema or configuration language structure. Deployments running previous >v3.1.x versions are strongly recommended to be upgraded to v3.1.4. > >For more details about version 3.1.4, visit: > >http://www.kamailio.org/w/2011/05/kamailio-v3-1-4-released/ > >Cheers, >Daniel > >-- >Daniel-Constantin Mierla -- http://www.asipto.com >http://linkedin.com/in/miconda -- http://twitter.com/miconda > > > > >------------------------------ > >Message: 4 >Date: Thu, 26 May 2011 17:59:42 -0300 >From: caio <elcaio(a)gmail.com> >Subject: Re: [SR-Users] ser_ctl / serweb >To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List" <sr-users(a)lists.sip-router.org> >Message-ID: <BANLkTik+dnTp-egK2xy1ua=tPnCcaDuuOA(a)mail.gmail.com> >Content-Type: text/plain; charset="utf-8" > >Hello, > >Anyone know if ser_ctl (python utility) will be merged into master? Or is >going to be deprecated? >If it still alive, are there any examples/docs of the usage for DB users >provisoining? >kamctl and kamdbctl are recomended for its substitution? > >Which utility will be official for sip-router? >Sorry if it already were discussed, but I didn't find comments about it >since 2010. > >Thank you >Claudio > >On Tue, May 24, 2011 at 11:14 AM, Claudio Furrer <elcaio(a)gmail.com> wrote: > >> Hi all, >> >> Just want to ask if ser_ctl application will be merged into sr3.1 or future >> releases. >> By the moment I've found it here [1], [2] or [3]. >> >> Is it full compatible with sip-router v3.x? >> >> I know kamctl and kamdbctl would be suggested but these are for kamailio >> flavour. I don't find too much work based on SER flavour regarding to >> database/users administration and web provisioning. >> BTW, siremis only works with kamailio db structures, but for sip-router v3 >> ser-flavoured only found serweb 2.x here [4] and [5] which I don't know if >> is >> it working well with the current new releases of the project. >> >> I would appreciate if someone can advice me if kamailio is the way to go, >> because have find that sip-router or ser flavours miss some docs and apps >> that >> kamailio already have. >> >> I'm asking from a a SER-user point of view, who want to migrate to SR-3 :( >> >> [1] http://ftp.iptel.org/pub/serctl/daily-snapshots/ >> [2] >> >> http://git.sip-router.org/cgi-bin/gitweb.cgi?p=ser;a=tree;f=ser_ctl;h=7db4a… >> [3] http://cvs.berlios.de/viewvc/ser/serctl/ >> >> [4] http://ftp.iptel.org/pub/serweb/daily-snapshots/ >> [5] http://developer.berlios.de/projects/serweb/ >> >> Thank you, >> Claudio >> >

1 0

Kamailio v3.1.4 Released
by Daniel-Constantin Mierla 26 May '11

26 May '11

Hello, Kamailio SIP Server v3.1.4 stable release is out. This is a maintenance release of latest stable branch, 3.1, that includes fixes since release of v3.1.3. There is no change to database schema or configuration language structure. Deployments running previous v3.1.x versions are strongly recommended to be upgraded to v3.1.4. For more details about version 3.1.4, visit: http://www.kamailio.org/w/2011/05/kamailio-v3-1-4-released/ Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

Aliases Using the Default Script;
by David J. 26 May '11

26 May '11

I am using the kamailio default script on 3.1.3. I was wondering what happens when I added an Alias in dbaliases? For example if I add 18005551212(a)mydomain.com alias to 1001(a)mydomain.com when an invite comes in; it works perfect I got a 200 back. (1001 Device rings.) If I add another alias to a remote server; ie. 18005551213(a)mydomain.com alias to 1800555124(a)someotherdomain.com I get a 404 back. I wonder what happens internally? What should I modify to enable this case; Thanks.

1 0

preparing v3.1.4
by Daniel-Constantin Mierla 26 May '11

26 May '11

Later today I will package v3.1.4, so if you have something to commit in branch 3.1, please do it before 15:00GMT. Thanks, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 0

packaging 3.1.4
by Juha Heinanen 26 May '11

26 May '11

Daniel-Constantin Mierla writes: > I am thinking of packaging v3.1.4 next week on Thursday -- last minor > release was more than 1.5month ago. > Any comments against it? the bug still exists, where AVP_CLASS_DOMAIN avps are not saved in the transaction. i guess jan forgot about it. -- juha

2 1

"ACK not received"
by Mokhtar Bengana 26 May '11

26 May '11

I am trying to forward a DID to a sip account registered to Kamailio. The call get established with two way audio and Kamailio shows it's comming from an Asterisk server. However the call get disconnected each time after 31s. Ngrep is showing "ACK not received" by the Kamailio test account before the call get disconnected. If Kamailio is suppose to send an ACK back to the asterisk server to keep the call up, I am not sure why it's not happening. Or is it the other way around? I have included the ngrep output. I appreciate any help. U 2011/05/26 00:27:12.525344 91.121.81.212:5060 -> 192.168.2.4:5060 INVITE sip:test@sipsoft.dyndns.org SIP/2.0. Record-Route: <sip:91.121.81.212;lr;mr=yes>. Via: SIP/2.0/UDP 91.121.81.212;branch=z9hG4bK33ad.da5ada05.0. Via: SIP/2.0/UDP 188.165.98.47:5060;branch=z9hG4bK60ba8b25;rport=5060. From: "3313144859493" <sip:3313144859493@188.165.98.47>;tag=as6c63c119. To: <sip:33980307151@sip.antisip.com:5060>. Contact: <sip:3313144859493@188.165.98.47:5060>. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 102 INVITE. User-Agent: Asterisk PBX. Max-Forwards: 69. Date: Thu, 26 May 2011 00:27:10 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces. Content-Type: application/sdp. Content-Length: 260. P-Antisip: forwarded with alias. P-PMR: mediarelay enabled. . v=0. o=root 10502 10502 IN IP4 91.121.81.212. s=session. c=IN IP4 91.121.81.212. t=0 0. m=audio 41316 RTP/AVP 8 101. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. a=sendrecv. a=nortpproxy:yes. U 2011/05/26 00:27:12.526734 192.168.2.4:5060 -> 91.121.81.212:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 91.121.81.212;branch=z9hG4bK33ad.da5ada05.0. Via: SIP/2.0/UDP 188.165.98.47:5060;branch=z9hG4bK60ba8b25;rport=5060. From: "3313144859493" <sip:3313144859493@188.165.98.47>;tag=as6c63c119. To: <sip:33980307151@sip.antisip.com:5060>. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 102 INVITE. Server: kamailio (3.1.3 (x86_64/linux)). Content-Length: 0. . U 2011/05/26 00:27:12.527013 192.168.2.4:5060 -> 192.168.2.8:65412 INVITE sip:test@192.168.2.8:65412;rinstance=befda46036a026ac SIP/2.0. Record-Route: <sip:192.168.2.4;lr=on>. Record-Route: <sip:91.121.81.212;lr;mr=yes>. Via: SIP/2.0/UDP 192.168.2.4;branch=z9hG4bK33ad.55f7f307.0. Via: SIP/2.0/UDP 91.121.81.212;branch=z9hG4bK33ad.da5ada05.0. Via: SIP/2.0/UDP 188.165.98.47:5060;branch=z9hG4bK60ba8b25;rport=5060. From: "3313144859493" <sip:3313144859493@188.165.98.47>;tag=as6c63c119. To: <sip:33980307151@sip.antisip.com:5060>. Contact: <sip:3313144859493@188.165.98.47:5060>. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 102 INVITE. User-Agent: Asterisk PBX. Max-Forwards: 68. Date: Thu, 26 May 2011 00:27:10 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces. Content-Type: application/sdp. Content-Length: 260. P-Antisip: forwarded with alias. P-PMR: mediarelay enabled. . v=0. o=root 10502 10502 IN IP4 91.121.81.212. s=session. c=IN IP4 91.121.81.212. t=0 0. m=audio 41316 RTP/AVP 8 101. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. a=sendrecv. a=nortpproxy:yes. U 2011/05/26 00:27:12.601815 192.168.2.8:65412 -> 192.168.2.4:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 192.168.2.4;branch=z9hG4bK33ad.55f7f307.0. Via: SIP/2.0/UDP 91.121.81.212;branch=z9hG4bK33ad.da5ada05.0. Via: SIP/2.0/UDP 188.165.98.47:5060;branch=z9hG4bK60ba8b25;rport=5060. Record-Route: <sip:192.168.2.4;lr>. Record-Route: <sip:91.121.81.212;lr;mr=yes>. Contact: <sip:test@192.168.2.8:65412;rinstance=befda46036a026ac>. To: <sip:33980307151@sip.antisip.com:5060>;tag=1df26a5d. From: "3313144859493"<sip:3313144859493@188.165.98.47>;tag=as6c63c119. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 102 INVITE. User-Agent: X-Lite 4 release 4.0 stamp 58832. Content-Length: 0. . U 2011/05/26 00:27:12.601967 192.168.2.4:5060 -> 91.121.81.212:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 91.121.81.212;branch=z9hG4bK33ad.da5ada05.0. Via: SIP/2.0/UDP 188.165.98.47:5060;branch=z9hG4bK60ba8b25;rport=5060. Record-Route: <sip:192.168.2.4;lr>. Record-Route: <sip:91.121.81.212;lr;mr=yes>. Contact: <sip:test@192.168.2.8:65412;rinstance=befda46036a026ac>. To: <sip:33980307151@sip.antisip.com:5060>;tag=1df26a5d. From: "3313144859493"<sip:3313144859493@188.165.98.47>;tag=as6c63c119. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 102 INVITE. User-Agent: X-Lite 4 release 4.0 stamp 58832. Content-Length: 0. . U 2011/05/26 00:27:22.568944 192.168.2.8:65412 -> 192.168.2.4:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.2.4;branch=z9hG4bK33ad.55f7f307.0. Via: SIP/2.0/UDP 91.121.81.212;branch=z9hG4bK33ad.da5ada05.0. Via: SIP/2.0/UDP 188.165.98.47:5060;branch=z9hG4bK60ba8b25;rport=5060. Record-Route: <sip:192.168.2.4;lr>. Record-Route: <sip:91.121.81.212;lr;mr=yes>. Contact: <sip:test@192.168.2.8:65412;rinstance=befda46036a026ac>. To: <sip:33980307151@sip.antisip.com:5060>;tag=1df26a5d. From: "3313144859493"<sip:3313144859493@188.165.98.47>;tag=as6c63c119. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 102 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite 4 release 4.0 stamp 58832. Content-Length: 371. . v=0. o=- 12950843244293000 1 IN IP4 192.168.2.8. s=CounterPath X-Lite 4.0. c=IN IP4 192.168.2.8. t=0 0. a=ice-ufrag:b5d539. a=ice-pwd:00ac4defa17299dce44e27bf54408a4c. m=audio 54492 RTP/AVP 8 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=candidate:1 1 UDP 659136 192.168.2.8 54492 typ host. a=candidate:1 2 UDP 659134 192.168.2.8 54493 typ host. U 2011/05/26 00:27:22.569043 192.168.2.4:5060 -> 91.121.81.212:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 91.121.81.212;branch=z9hG4bK33ad.da5ada05.0. Via: SIP/2.0/UDP 188.165.98.47:5060;branch=z9hG4bK60ba8b25;rport=5060. Record-Route: <sip:192.168.2.4;lr>. Record-Route: <sip:91.121.81.212;lr;mr=yes>. Contact: <sip:test@192.168.2.8:65412;rinstance=befda46036a026ac>. To: <sip:33980307151@sip.antisip.com:5060>;tag=1df26a5d. From: "3313144859493"<sip:3313144859493@188.165.98.47>;tag=as6c63c119. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 102 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite 4 release 4.0 stamp 58832. Content-Length: 371. . v=0. o=- 12950843244293000 1 IN IP4 192.168.2.8. s=CounterPath X-Lite 4.0. c=IN IP4 192.168.2.8. t=0 0. a=ice-ufrag:b5d539. a=ice-pwd:00ac4defa17299dce44e27bf54408a4c. m=audio 54492 RTP/AVP 8 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=candidate:1 1 UDP 659136 192.168.2.8 54492 typ host. a=candidate:1 2 UDP 659134 192.168.2.8 54493 typ host. U 2011/05/26 00:27:23.080398 192.168.2.8:65412 -> 192.168.2.4:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.2.4;branch=z9hG4bK33ad.55f7f307.0. Via: SIP/2.0/UDP 91.121.81.212;branch=z9hG4bK33ad.da5ada05.0. Via: SIP/2.0/UDP 188.165.98.47:5060;branch=z9hG4bK60ba8b25;rport=5060. Record-Route: <sip:192.168.2.4;lr>. Record-Route: <sip:91.121.81.212;lr;mr=yes>. Contact: <sip:test@192.168.2.8:65412;rinstance=befda46036a026ac>. To: <sip:33980307151@sip.antisip.com:5060>;tag=1df26a5d. From: "3313144859493"<sip:3313144859493@188.165.98.47>;tag=as6c63c119. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 102 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite 4 release 4.0 stamp 58832. Content-Length: 371. . v=0. o=- 12950843244293000 1 IN IP4 192.168.2.8. s=CounterPath X-Lite 4.0. c=IN IP4 192.168.2.8. t=0 0. a=ice-ufrag:b5d539. a=ice-pwd:00ac4defa17299dce44e27bf54408a4c. m=audio 54492 RTP/AVP 8 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=candidate:1 1 UDP 659136 192.168.2.8 54492 typ host. a=candidate:1 2 UDP 659134 192.168.2.8 54493 typ host. U 2011/05/26 00:27:23.080549 192.168.2.4:5060 -> 91.121.81.212:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 91.121.81.212;branch=z9hG4bK33ad.da5ada05.0. Via: SIP/2.0/UDP 188.165.98.47:5060;branch=z9hG4bK60ba8b25;rport=5060. Record-Route: <sip:192.168.2.4;lr>. Record-Route: <sip:91.121.81.212;lr;mr=yes>. Contact: <sip:test@192.168.2.8:65412;rinstance=befda46036a026ac>. To: <sip:33980307151@sip.antisip.com:5060>;tag=1df26a5d. From: "3313144859493"<sip:3313144859493@188.165.98.47>;tag=as6c63c119. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 102 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite 4 release 4.0 stamp 58832. Content-Length: 371. . v=0. o=- 12950843244293000 1 IN IP4 192.168.2.8. s=CounterPath X-Lite 4.0. c=IN IP4 192.168.2.8. t=0 0. a=ice-ufrag:b5d539. a=ice-pwd:00ac4defa17299dce44e27bf54408a4c. m=audio 54492 RTP/AVP 8 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=candidate:1 1 UDP 659136 192.168.2.8 54492 typ host. a=candidate:1 2 UDP 659134 192.168.2.8 54493 typ host. U 2011/05/26 00:27:24.080628 192.168.2.8:65412 -> 192.168.2.4:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.2.4;branch=z9hG4bK33ad.55f7f307.0. Via: SIP/2.0/UDP 91.121.81.212;branch=z9hG4bK33ad.da5ada05.0. Via: SIP/2.0/UDP 188.165.98.47:5060;branch=z9hG4bK60ba8b25;rport=5060. Record-Route: <sip:192.168.2.4;lr>. Record-Route: <sip:91.121.81.212;lr;mr=yes>. Contact: <sip:test@192.168.2.8:65412;rinstance=befda46036a026ac>. To: <sip:33980307151@sip.antisip.com:5060>;tag=1df26a5d. From: "3313144859493"<sip:3313144859493@188.165.98.47>;tag=as6c63c119. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 102 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite 4 release 4.0 stamp 58832. Content-Length: 371. . v=0. o=- 12950843244293000 1 IN IP4 192.168.2.8. s=CounterPath X-Lite 4.0. c=IN IP4 192.168.2.8. t=0 0. a=ice-ufrag:b5d539. a=ice-pwd:00ac4defa17299dce44e27bf54408a4c. m=audio 54492 RTP/AVP 8 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=candidate:1 1 UDP 659136 192.168.2.8 54492 typ host. a=candidate:1 2 UDP 659134 192.168.2.8 54493 typ host. U 2011/05/26 00:27:24.080697 192.168.2.4:5060 -> 91.121.81.212:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 91.121.81.212;branch=z9hG4bK33ad.da5ada05.0. Via: SIP/2.0/UDP 188.165.98.47:5060;branch=z9hG4bK60ba8b25;rport=5060. Record-Route: <sip:192.168.2.4;lr>. Record-Route: <sip:91.121.81.212;lr;mr=yes>. Contact: <sip:test@192.168.2.8:65412;rinstance=befda46036a026ac>. To: <sip:33980307151@sip.antisip.com:5060>;tag=1df26a5d. From: "3313144859493"<sip:3313144859493@188.165.98.47>;tag=as6c63c119. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 102 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite 4 release 4.0 stamp 58832. Content-Length: 371. . v=0. o=- 12950843244293000 1 IN IP4 192.168.2.8. s=CounterPath X-Lite 4.0. c=IN IP4 192.168.2.8. t=0 0. a=ice-ufrag:b5d539. a=ice-pwd:00ac4defa17299dce44e27bf54408a4c. m=audio 54492 RTP/AVP 8 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=candidate:1 1 UDP 659136 192.168.2.8 54492 typ host. a=candidate:1 2 UDP 659134 192.168.2.8 54493 typ host. U 2011/05/26 00:27:26.080654 192.168.2.8:65412 -> 192.168.2.4:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.2.4;branch=z9hG4bK33ad.55f7f307.0. Via: SIP/2.0/UDP 91.121.81.212;branch=z9hG4bK33ad.da5ada05.0. Via: SIP/2.0/UDP 188.165.98.47:5060;branch=z9hG4bK60ba8b25;rport=5060. Record-Route: <sip:192.168.2.4;lr>. Record-Route: <sip:91.121.81.212;lr;mr=yes>. Contact: <sip:test@192.168.2.8:65412;rinstance=befda46036a026ac>. To: <sip:33980307151@sip.antisip.com:5060>;tag=1df26a5d. From: "3313144859493"<sip:3313144859493@188.165.98.47>;tag=as6c63c119. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 102 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite 4 release 4.0 stamp 58832. Content-Length: 371. . v=0. o=- 12950843244293000 1 IN IP4 192.168.2.8. s=CounterPath X-Lite 4.0. c=IN IP4 192.168.2.8. t=0 0. a=ice-ufrag:b5d539. a=ice-pwd:00ac4defa17299dce44e27bf54408a4c. m=audio 54492 RTP/AVP 8 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=candidate:1 1 UDP 659136 192.168.2.8 54492 typ host. a=candidate:1 2 UDP 659134 192.168.2.8 54493 typ host. U 2011/05/26 00:27:26.080816 192.168.2.4:5060 -> 91.121.81.212:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 91.121.81.212;branch=z9hG4bK33ad.da5ada05.0. Via: SIP/2.0/UDP 188.165.98.47:5060;branch=z9hG4bK60ba8b25;rport=5060. Record-Route: <sip:192.168.2.4;lr>. Record-Route: <sip:91.121.81.212;lr;mr=yes>. Contact: <sip:test@192.168.2.8:65412;rinstance=befda46036a026ac>. To: <sip:33980307151@sip.antisip.com:5060>;tag=1df26a5d. From: "3313144859493"<sip:3313144859493@188.165.98.47>;tag=as6c63c119. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 102 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite 4 release 4.0 stamp 58832. Content-Length: 371. . v=0. o=- 12950843244293000 1 IN IP4 192.168.2.8. s=CounterPath X-Lite 4.0. c=IN IP4 192.168.2.8. t=0 0. a=ice-ufrag:b5d539. a=ice-pwd:00ac4defa17299dce44e27bf54408a4c. m=audio 54492 RTP/AVP 8 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=candidate:1 1 UDP 659136 192.168.2.8 54492 typ host. a=candidate:1 2 UDP 659134 192.168.2.8 54493 typ host. U 2011/05/26 00:27:30.080527 192.168.2.8:65412 -> 192.168.2.4:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.2.4;branch=z9hG4bK33ad.55f7f307.0. Via: SIP/2.0/UDP 91.121.81.212;branch=z9hG4bK33ad.da5ada05.0. Via: SIP/2.0/UDP 188.165.98.47:5060;branch=z9hG4bK60ba8b25;rport=5060. Record-Route: <sip:192.168.2.4;lr>. Record-Route: <sip:91.121.81.212;lr;mr=yes>. Contact: <sip:test@192.168.2.8:65412;rinstance=befda46036a026ac>. To: <sip:33980307151@sip.antisip.com:5060>;tag=1df26a5d. From: "3313144859493"<sip:3313144859493@188.165.98.47>;tag=as6c63c119. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 102 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite 4 release 4.0 stamp 58832. Content-Length: 371. . v=0. o=- 12950843244293000 1 IN IP4 192.168.2.8. s=CounterPath X-Lite 4.0. c=IN IP4 192.168.2.8. t=0 0. a=ice-ufrag:b5d539. a=ice-pwd:00ac4defa17299dce44e27bf54408a4c. m=audio 54492 RTP/AVP 8 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=candidate:1 1 UDP 659136 192.168.2.8 54492 typ host. a=candidate:1 2 UDP 659134 192.168.2.8 54493 typ host. U 2011/05/26 00:27:30.080594 192.168.2.4:5060 -> 91.121.81.212:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 91.121.81.212;branch=z9hG4bK33ad.da5ada05.0. Via: SIP/2.0/UDP 188.165.98.47:5060;branch=z9hG4bK60ba8b25;rport=5060. Record-Route: <sip:192.168.2.4;lr>. Record-Route: <sip:91.121.81.212;lr;mr=yes>. Contact: <sip:test@192.168.2.8:65412;rinstance=befda46036a026ac>. To: <sip:33980307151@sip.antisip.com:5060>;tag=1df26a5d. From: "3313144859493"<sip:3313144859493@188.165.98.47>;tag=as6c63c119. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 102 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite 4 release 4.0 stamp 58832. Content-Length: 371. . v=0. o=- 12950843244293000 1 IN IP4 192.168.2.8. s=CounterPath X-Lite 4.0. c=IN IP4 192.168.2.8. t=0 0. a=ice-ufrag:b5d539. a=ice-pwd:00ac4defa17299dce44e27bf54408a4c. m=audio 54492 RTP/AVP 8 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=candidate:1 1 UDP 659136 192.168.2.8 54492 typ host. a=candidate:1 2 UDP 659134 192.168.2.8 54493 typ host. U 2011/05/26 00:27:34.080318 192.168.2.8:65412 -> 192.168.2.4:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.2.4;branch=z9hG4bK33ad.55f7f307.0. Via: SIP/2.0/UDP 91.121.81.212;branch=z9hG4bK33ad.da5ada05.0. Via: SIP/2.0/UDP 188.165.98.47:5060;branch=z9hG4bK60ba8b25;rport=5060. Record-Route: <sip:192.168.2.4;lr>. Record-Route: <sip:91.121.81.212;lr;mr=yes>. Contact: <sip:test@192.168.2.8:65412;rinstance=befda46036a026ac>. To: <sip:33980307151@sip.antisip.com:5060>;tag=1df26a5d. From: "3313144859493"<sip:3313144859493@188.165.98.47>;tag=as6c63c119. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 102 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite 4 release 4.0 stamp 58832. Content-Length: 371. . v=0. o=- 12950843244293000 1 IN IP4 192.168.2.8. s=CounterPath X-Lite 4.0. c=IN IP4 192.168.2.8. t=0 0. a=ice-ufrag:b5d539. a=ice-pwd:00ac4defa17299dce44e27bf54408a4c. m=audio 54492 RTP/AVP 8 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=candidate:1 1 UDP 659136 192.168.2.8 54492 typ host. a=candidate:1 2 UDP 659134 192.168.2.8 54493 typ host. U 2011/05/26 00:27:34.080436 192.168.2.4:5060 -> 91.121.81.212:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 91.121.81.212;branch=z9hG4bK33ad.da5ada05.0. Via: SIP/2.0/UDP 188.165.98.47:5060;branch=z9hG4bK60ba8b25;rport=5060. Record-Route: <sip:192.168.2.4;lr>. Record-Route: <sip:91.121.81.212;lr;mr=yes>. Contact: <sip:test@192.168.2.8:65412;rinstance=befda46036a026ac>. To: <sip:33980307151@sip.antisip.com:5060>;tag=1df26a5d. From: "3313144859493"<sip:3313144859493@188.165.98.47>;tag=as6c63c119. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 102 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite 4 release 4.0 stamp 58832. Content-Length: 371. . v=0. o=- 12950843244293000 1 IN IP4 192.168.2.8. s=CounterPath X-Lite 4.0. c=IN IP4 192.168.2.8. t=0 0. a=ice-ufrag:b5d539. a=ice-pwd:00ac4defa17299dce44e27bf54408a4c. m=audio 54492 RTP/AVP 8 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=candidate:1 1 UDP 659136 192.168.2.8 54492 typ host. a=candidate:1 2 UDP 659134 192.168.2.8 54493 typ host. U 2011/05/26 00:27:37.137775 192.168.2.8:65412 -> 192.168.2.4:5060 . . U 2011/05/26 00:27:38.081552 192.168.2.8:65412 -> 192.168.2.4:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.2.4;branch=z9hG4bK33ad.55f7f307.0. Via: SIP/2.0/UDP 91.121.81.212;branch=z9hG4bK33ad.da5ada05.0. Via: SIP/2.0/UDP 188.165.98.47:5060;branch=z9hG4bK60ba8b25;rport=5060. Record-Route: <sip:192.168.2.4;lr>. Record-Route: <sip:91.121.81.212;lr;mr=yes>. Contact: <sip:test@192.168.2.8:65412;rinstance=befda46036a026ac>. To: <sip:33980307151@sip.antisip.com:5060>;tag=1df26a5d. From: "3313144859493"<sip:3313144859493@188.165.98.47>;tag=as6c63c119. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 102 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite 4 release 4.0 stamp 58832. Content-Length: 371. . v=0. o=- 12950843244293000 1 IN IP4 192.168.2.8. s=CounterPath X-Lite 4.0. c=IN IP4 192.168.2.8. t=0 0. a=ice-ufrag:b5d539. a=ice-pwd:00ac4defa17299dce44e27bf54408a4c. m=audio 54492 RTP/AVP 8 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=candidate:1 1 UDP 659136 192.168.2.8 54492 typ host. a=candidate:1 2 UDP 659134 192.168.2.8 54493 typ host. U 2011/05/26 00:27:38.081707 192.168.2.4:5060 -> 91.121.81.212:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 91.121.81.212;branch=z9hG4bK33ad.da5ada05.0. Via: SIP/2.0/UDP 188.165.98.47:5060;branch=z9hG4bK60ba8b25;rport=5060. Record-Route: <sip:192.168.2.4;lr>. Record-Route: <sip:91.121.81.212;lr;mr=yes>. Contact: <sip:test@192.168.2.8:65412;rinstance=befda46036a026ac>. To: <sip:33980307151@sip.antisip.com:5060>;tag=1df26a5d. From: "3313144859493"<sip:3313144859493@188.165.98.47>;tag=as6c63c119. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 102 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite 4 release 4.0 stamp 58832. Content-Length: 371. . v=0. o=- 12950843244293000 1 IN IP4 192.168.2.8. s=CounterPath X-Lite 4.0. c=IN IP4 192.168.2.8. t=0 0. a=ice-ufrag:b5d539. a=ice-pwd:00ac4defa17299dce44e27bf54408a4c. m=audio 54492 RTP/AVP 8 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=candidate:1 1 UDP 659136 192.168.2.8 54492 typ host. a=candidate:1 2 UDP 659134 192.168.2.8 54493 typ host. U 2011/05/26 00:27:42.082163 192.168.2.8:65412 -> 192.168.2.4:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.2.4;branch=z9hG4bK33ad.55f7f307.0. Via: SIP/2.0/UDP 91.121.81.212;branch=z9hG4bK33ad.da5ada05.0. Via: SIP/2.0/UDP 188.165.98.47:5060;branch=z9hG4bK60ba8b25;rport=5060. Record-Route: <sip:192.168.2.4;lr>. Record-Route: <sip:91.121.81.212;lr;mr=yes>. Contact: <sip:test@192.168.2.8:65412;rinstance=befda46036a026ac>. To: <sip:33980307151@sip.antisip.com:5060>;tag=1df26a5d. From: "3313144859493"<sip:3313144859493@188.165.98.47>;tag=as6c63c119. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 102 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite 4 release 4.0 stamp 58832. Content-Length: 371. . v=0. o=- 12950843244293000 1 IN IP4 192.168.2.8. s=CounterPath X-Lite 4.0. c=IN IP4 192.168.2.8. t=0 0. a=ice-ufrag:b5d539. a=ice-pwd:00ac4defa17299dce44e27bf54408a4c. m=audio 54492 RTP/AVP 8 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=candidate:1 1 UDP 659136 192.168.2.8 54492 typ host. a=candidate:1 2 UDP 659134 192.168.2.8 54493 typ host. U 2011/05/26 00:27:42.082308 192.168.2.4:5060 -> 91.121.81.212:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 91.121.81.212;branch=z9hG4bK33ad.da5ada05.0. Via: SIP/2.0/UDP 188.165.98.47:5060;branch=z9hG4bK60ba8b25;rport=5060. Record-Route: <sip:192.168.2.4;lr>. Record-Route: <sip:91.121.81.212;lr;mr=yes>. Contact: <sip:test@192.168.2.8:65412;rinstance=befda46036a026ac>. To: <sip:33980307151@sip.antisip.com:5060>;tag=1df26a5d. From: "3313144859493"<sip:3313144859493@188.165.98.47>;tag=as6c63c119. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 102 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite 4 release 4.0 stamp 58832. Content-Length: 371. . v=0. o=- 12950843244293000 1 IN IP4 192.168.2.8. s=CounterPath X-Lite 4.0. c=IN IP4 192.168.2.8. t=0 0. a=ice-ufrag:b5d539. a=ice-pwd:00ac4defa17299dce44e27bf54408a4c. m=audio 54492 RTP/AVP 8 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=candidate:1 1 UDP 659136 192.168.2.8 54492 typ host. a=candidate:1 2 UDP 659134 192.168.2.8 54493 typ host. U 2011/05/26 00:27:46.080414 192.168.2.8:65412 -> 192.168.2.4:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.2.4;branch=z9hG4bK33ad.55f7f307.0. Via: SIP/2.0/UDP 91.121.81.212;branch=z9hG4bK33ad.da5ada05.0. Via: SIP/2.0/UDP 188.165.98.47:5060;branch=z9hG4bK60ba8b25;rport=5060. Record-Route: <sip:192.168.2.4;lr>. Record-Route: <sip:91.121.81.212;lr;mr=yes>. Contact: <sip:test@192.168.2.8:65412;rinstance=befda46036a026ac>. To: <sip:33980307151@sip.antisip.com:5060>;tag=1df26a5d. From: "3313144859493"<sip:3313144859493@188.165.98.47>;tag=as6c63c119. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 102 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite 4 release 4.0 stamp 58832. Content-Length: 371. . v=0. o=- 12950843244293000 1 IN IP4 192.168.2.8. s=CounterPath X-Lite 4.0. c=IN IP4 192.168.2.8. t=0 0. a=ice-ufrag:b5d539. a=ice-pwd:00ac4defa17299dce44e27bf54408a4c. m=audio 54492 RTP/AVP 8 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=candidate:1 1 UDP 659136 192.168.2.8 54492 typ host. a=candidate:1 2 UDP 659134 192.168.2.8 54493 typ host. U 2011/05/26 00:27:46.080562 192.168.2.4:5060 -> 91.121.81.212:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 91.121.81.212;branch=z9hG4bK33ad.da5ada05.0. Via: SIP/2.0/UDP 188.165.98.47:5060;branch=z9hG4bK60ba8b25;rport=5060. Record-Route: <sip:192.168.2.4;lr>. Record-Route: <sip:91.121.81.212;lr;mr=yes>. Contact: <sip:test@192.168.2.8:65412;rinstance=befda46036a026ac>. To: <sip:33980307151@sip.antisip.com:5060>;tag=1df26a5d. From: "3313144859493"<sip:3313144859493@188.165.98.47>;tag=as6c63c119. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 102 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite 4 release 4.0 stamp 58832. Content-Length: 371. . v=0. o=- 12950843244293000 1 IN IP4 192.168.2.8. s=CounterPath X-Lite 4.0. c=IN IP4 192.168.2.8. t=0 0. a=ice-ufrag:b5d539. a=ice-pwd:00ac4defa17299dce44e27bf54408a4c. m=audio 54492 RTP/AVP 8 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=candidate:1 1 UDP 659136 192.168.2.8 54492 typ host. a=candidate:1 2 UDP 659134 192.168.2.8 54493 typ host. U 2011/05/26 00:27:50.082011 192.168.2.8:65412 -> 192.168.2.4:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.2.4;branch=z9hG4bK33ad.55f7f307.0. Via: SIP/2.0/UDP 91.121.81.212;branch=z9hG4bK33ad.da5ada05.0. Via: SIP/2.0/UDP 188.165.98.47:5060;branch=z9hG4bK60ba8b25;rport=5060. Record-Route: <sip:192.168.2.4;lr>. Record-Route: <sip:91.121.81.212;lr;mr=yes>. Contact: <sip:test@192.168.2.8:65412;rinstance=befda46036a026ac>. To: <sip:33980307151@sip.antisip.com:5060>;tag=1df26a5d. From: "3313144859493"<sip:3313144859493@188.165.98.47>;tag=as6c63c119. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 102 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite 4 release 4.0 stamp 58832. Content-Length: 371. . v=0. o=- 12950843244293000 1 IN IP4 192.168.2.8. s=CounterPath X-Lite 4.0. c=IN IP4 192.168.2.8. t=0 0. a=ice-ufrag:b5d539. a=ice-pwd:00ac4defa17299dce44e27bf54408a4c. m=audio 54492 RTP/AVP 8 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=candidate:1 1 UDP 659136 192.168.2.8 54492 typ host. a=candidate:1 2 UDP 659134 192.168.2.8 54493 typ host. U 2011/05/26 00:27:50.082142 192.168.2.4:5060 -> 91.121.81.212:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 91.121.81.212;branch=z9hG4bK33ad.da5ada05.0. Via: SIP/2.0/UDP 188.165.98.47:5060;branch=z9hG4bK60ba8b25;rport=5060. Record-Route: <sip:192.168.2.4;lr>. Record-Route: <sip:91.121.81.212;lr;mr=yes>. Contact: <sip:test@192.168.2.8:65412;rinstance=befda46036a026ac>. To: <sip:33980307151@sip.antisip.com:5060>;tag=1df26a5d. From: "3313144859493"<sip:3313144859493@188.165.98.47>;tag=as6c63c119. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 102 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite 4 release 4.0 stamp 58832. Content-Length: 371. . v=0. o=- 12950843244293000 1 IN IP4 192.168.2.8. s=CounterPath X-Lite 4.0. c=IN IP4 192.168.2.8. t=0 0. a=ice-ufrag:b5d539. a=ice-pwd:00ac4defa17299dce44e27bf54408a4c. m=audio 54492 RTP/AVP 8 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=candidate:1 1 UDP 659136 192.168.2.8 54492 typ host. a=candidate:1 2 UDP 659134 192.168.2.8 54493 typ host. U 2011/05/26 00:27:54.080488 192.168.2.8:65412 -> 192.168.2.4:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.2.4;branch=z9hG4bK33ad.55f7f307.0. Via: SIP/2.0/UDP 91.121.81.212;branch=z9hG4bK33ad.da5ada05.0. Via: SIP/2.0/UDP 188.165.98.47:5060;branch=z9hG4bK60ba8b25;rport=5060. Record-Route: <sip:192.168.2.4;lr>. Record-Route: <sip:91.121.81.212;lr;mr=yes>. Contact: <sip:test@192.168.2.8:65412;rinstance=befda46036a026ac>. To: <sip:33980307151@sip.antisip.com:5060>;tag=1df26a5d. From: "3313144859493"<sip:3313144859493@188.165.98.47>;tag=as6c63c119. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 102 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite 4 release 4.0 stamp 58832. Content-Length: 371. . v=0. o=- 12950843244293000 1 IN IP4 192.168.2.8. s=CounterPath X-Lite 4.0. c=IN IP4 192.168.2.8. t=0 0. a=ice-ufrag:b5d539. a=ice-pwd:00ac4defa17299dce44e27bf54408a4c. m=audio 54492 RTP/AVP 8 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=candidate:1 1 UDP 659136 192.168.2.8 54492 typ host. a=candidate:1 2 UDP 659134 192.168.2.8 54493 typ host. U 2011/05/26 00:27:54.080592 192.168.2.4:5060 -> 91.121.81.212:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 91.121.81.212;branch=z9hG4bK33ad.da5ada05.0. Via: SIP/2.0/UDP 188.165.98.47:5060;branch=z9hG4bK60ba8b25;rport=5060. Record-Route: <sip:192.168.2.4;lr>. Record-Route: <sip:91.121.81.212;lr;mr=yes>. Contact: <sip:test@192.168.2.8:65412;rinstance=befda46036a026ac>. To: <sip:33980307151@sip.antisip.com:5060>;tag=1df26a5d. From: "3313144859493"<sip:3313144859493@188.165.98.47>;tag=as6c63c119. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 102 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces. User-Agent: X-Lite 4 release 4.0 stamp 58832. Content-Length: 371. . v=0. o=- 12950843244293000 1 IN IP4 192.168.2.8. s=CounterPath X-Lite 4.0. c=IN IP4 192.168.2.8. t=0 0. a=ice-ufrag:b5d539. a=ice-pwd:00ac4defa17299dce44e27bf54408a4c. m=audio 54492 RTP/AVP 8 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. a=candidate:1 1 UDP 659136 192.168.2.8 54492 typ host. a=candidate:1 2 UDP 659134 192.168.2.8 54493 typ host. U 2011/05/26 00:27:54.574347 192.168.2.8:65412 -> 192.168.2.4:5060 BYE sip:3313144859493@188.165.98.47:5060 SIP/2.0. Via: SIP/2.0/UDP 192.168.2.8:65412;branch=z9hG4bK-d8754z-6e34d343af083241-1---d8754z-;rport. Max-Forwards: 70. Route: <sip:192.168.2.4;lr>. Route: <sip:91.121.81.212;lr;mr=yes>. Contact: <sip:test@192.168.2.8:65412;rinstance=befda46036a026ac>. To: "3313144859493"<sip:3313144859493@188.165.98.47>;tag=as6c63c119. From: <sip:33980307151@sip.antisip.com:5060>;tag=1df26a5d. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 2 BYE. User-Agent: X-Lite 4 release 4.0 stamp 58832. Reason: SIP;description="ACK not received". Content-Length: 0. . U 2011/05/26 00:27:54.574553 192.168.2.4:5060 -> 91.121.81.212:5060 BYE sip:3313144859493@188.165.98.47:5060 SIP/2.0. Via: SIP/2.0/UDP 192.168.2.4;branch=z9hG4bK79bb.32809f01.0. Via: SIP/2.0/UDP 192.168.2.8:65412;branch=z9hG4bK-d8754z-6e34d343af083241-1---d8754z-;rport=65412. Max-Forwards: 69. Route: <sip:91.121.81.212;lr;mr=yes>. Contact: <sip:test@192.168.2.8:65412;rinstance=befda46036a026ac>. To: "3313144859493"<sip:3313144859493@188.165.98.47>;tag=as6c63c119. From: <sip:33980307151@sip.antisip.com:5060>;tag=1df26a5d. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 2 BYE. User-Agent: X-Lite 4 release 4.0 stamp 58832. Reason: SIP;description="ACK not received". Content-Length: 0. . U 2011/05/26 00:27:54.705228 91.121.81.212:5060 -> 192.168.2.4:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.2.4;rport=5060;received=99.42.204.9;branch=z9hG4bK79bb.32809f01.0. Via: SIP/2.0/UDP 192.168.2.8:65412;branch=z9hG4bK-d8754z-6e34d343af083241-1---d8754z-;rport=65412. Record-Route: <sip:91.121.81.212;lr>. From: <sip:33980307151@sip.antisip.com:5060>;tag=1df26a5d. To: "3313144859493"<sip:3313144859493@188.165.98.47>;tag=as6c63c119. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 2 BYE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces. Content-Length: 0. . U 2011/05/26 00:27:54.705377 192.168.2.4:5060 -> 192.168.2.8:65412 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.2.8:65412;branch=z9hG4bK-d8754z-6e34d343af083241-1---d8754z-;rport=65412. Record-Route: <sip:91.121.81.212;lr>. From: <sip:33980307151@sip.antisip.com:5060>;tag=1df26a5d. To: "3313144859493"<sip:3313144859493@188.165.98.47>;tag=as6c63c119. Call-ID: 5e5ba375661c69273f32a463764879b0(a)188.165.98.47. CSeq: 2 BYE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces. Content-Length: 0. . U 2011/05/26 00:28:07.139142 192.168.2.8:65412 -> 192.168.2.4:5060 . .

3 3

Wildcard TLS
by Jon Farmer 25 May '11

25 May '11

Hi I am just wondering if it is possible to use a wildcard certificate with TLS. I have hundreds of customers who are using subdomains of the main domain name on a virtual PBX platform. i.e if the main domain is example.org I have customers using cust1.example.org cust2.example.org cust3.example.org ....... I want to start offering TLS on a per customer basis. However I don't want to have to create a TLS certificate per customer if I don't need to. So I need to know if I can create a *.example.org certificate and use that for all customers who want TLS? Is this possible? Regards Jon -- Jon Farmer Tel: 07795 118140 Email: viperdudeuk(a)gmail.com Twitter: @viperdudeuk

3 4

3.1.x and Kamailio compatibility mode
by Klaus Darilion 25 May '11

25 May '11

Hi! Is #!KAMAILIO still necessary with Kamailio 3.1.x? If yes, which behavior is changed by #!KAMAILIO? Thanks Klaus

3 8

Service Execution Time
by Giuseppe Carella 25 May '11

25 May '11

Hi all, I'm using kamailio with the dispatcher module. I would like to know if there are some API to get an average of the service execution time. In other words, I need to get the time between sending the request and receiving the response for every request, but also to have this values separate per server. I know also that this could be a little heavy for the dispatcher, but for some test I need this information. Your help is kindly appreciated! Thanks in advance, Giuseppe

2 1

Manipulate From header
by alex pappas 25 May '11

25 May '11

Dear all, I'm trying to change the From uri and Dsplay but without sucess. My config is as follow: if(dp_translate("1111", "$avp(s:frm_user_name)/$avp(s:test_frm_user_name)")) --> i'm sending 2112202701 and I get back corectly 701 { $avp(s:display) = $avp(s:test_frm_user_name); $avp(s:fu_uri) = "sip:" + $avp(s:test_frm_user_name) + "@" + $fd; xlog("alx ------- The avp(s:display): $avp(s:display) ---- avp(s:fu_uri)=$avp(s:fu_uri) ----------\n"); --> I see values 701 and sip:701@my_IP_address uac_replace_from("$avp(s:display)","$avp(s:fu_uri)"); Here I log the fu and I see that nothing has changed.. } Do you have any ideas on that ? I use Kamailio 3.1.3 Thank you in advance Alex

2 2

undefined symbol error
by Bruno Bresciani 24 May '11

24 May '11

Hi, Someone know tell me why kamailio generate the following error when I try load the route.so module: May 24 10:21:43 sswpst00 kamailio: ERROR: <core> [sr_module.c:523]: ERROR: load_module: could not open module </home2/local/kamailio/lib/kamailio/modules/route.so>: /home2/local/kamailio/lib/kamailio/modules/route.so: undefined symbol: db_use_table I don't know why this error have been generate. I'm porting this module to use with kamailio 3.1.2.

2 3

htable ::size has wrong value
by Juha Heinanen 24 May '11

24 May '11

i have entries in htable with key name lpi::20033::prefix: # sip-proxy_ctl mi sht_dump htable | egrep 20033 lpi::20033::prefix[0]:: +3583796294 lpi::20033::prefix[1]:: +35837824030 lpi::20033::prefix::size:: 1 as you see in above, size of that key name is 1 although there clearly is two entries in the table with that key name. according to readme ::size is number of items: 3.7. array_size_suffix (str) The suffix to be added to store the number of items in an array. i verified with another key name where there was only one prefix, and then ::size was again 1. so sometimes ::size is correct and sometimes incorrect. this looks like a serious bug to me. any comments? -- juha

2 4

Telefonia ip
by rom ignacio 24 May '11

24 May '11

friend I am interested in learning more about VoIP or IP telephony, I read a lot about you and I still have more to read, as it can not released through theofficial documentation page errors Please let me send them am email, thanks I DEVELOP voip server to sip in Venezuela, I need all the help possible. thanks *Rommel Malave .M ** * *T.S.U Universitario Extracción y Producción de Gas * * T.S.U en Informatica*** 04148241688

1 0

Re: [SR-Users] app_lua in kamailio
by Daniel-Constantin Mierla 24 May '11

24 May '11

Hello, please do not write direct private emails for questions about kamailio, use public mailing list sr-users(a)lists.sip-router.org (cc-ed) -- private emails are ignored, after the first notification. exit is not a function exported by a module, it is implemented in core. On the other hand, you try to stop the kamailio cfg interpreter when lua interpreter is executing the Lua script, so it won't work straightforward. Set a variable in kamailio from Lua, then check it after executing the lua script to decide to further exit or not. Cheers, Daniel On 5/24/11 11:44 AM, Bret McDanel wrote: > I am having a terrible time trying to get a conditional response from a > lua script. > > Basically I want > > if error then > sr.sl.send_reply(404,"Not Found"); > sr.modf("exit"); -- this does not work! > end > > > How can I have such a conditional? I have tried with return values from > the script to see if that would change anything but that did not seem to > work, sr.modf("exit") is not found. > > Any assistance would be greatly appreciated. > -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

3 3

struct run_act_ctx
by Bruno Bresciani 24 May '11

24 May '11

Hi, I'd like to know the functionality of the struct run_act_ctx. I need to adapt my route module to use this struct, the function do_action require this parameter. I'm migrating of the kamailio 1.5.0 to 3.1.2. Regards,

2 2

Re: [SR-Users] 403 Proxy forbidden replay for invite that send sip gateway
by hala alramli 24 May '11

24 May '11

thanks for your reply Klaus ,attached my config file could you please take a look ,i found the line that send that replay but i do not know why .? could you help me i am stuck --- Mer 18/5/11, Klaus Darilion <klaus.mailinglists(a)pernau.at> ha scritto: Da: Klaus Darilion <klaus.mailinglists(a)pernau.at> Oggetto: Re: [SR-Users] 403 Proxy forbidden replay for invite that send sip gateway A: sr-users(a)lists.sip-router.org Data: Mercoledì 18 maggio 2011, 13:04 Check your config and verify where 403 is sent. You have something like sl_send_reply("403","Proxy forbidden") in your config. regards Klaus Am 18.05.2011 10:48, schrieb hala alramli: > Hello every body, > could you help me ? > > i bought Geographical DID numbers from gateway provider , to receive > calls from pstn > > my problem that when gateway send invite my proxy send 403 forbidden ,i > dont know why what is wrong ,any help please. > > this is a track for th gateway invite. > > my proxy ip:79.12.254.153 > gateway IP :77.239.128.10 > my geographical number :0698962030 > > > > > > INVITE sip:0698962030@79.12.254.153:5060;user=phone SIP/2.0 > > Via: SIP/2.0/UDP 77.239.128.10:5060;branch=z9hG4bKf89p48m8dkuco1tumbnfaemc44 > > From: > <sip:0289089500@10.39.1.16:5060;user=phone>;tag=SDml08d01--45026-1f4d3a5-203bbf09-1f4d3a5 > > To: <sip:0698962030@10.39.6.207:5060;user=phone> > > Call-ID: SDml08d01-daae53b1c06b7367e53d68ad67a63d13-ao92061 > > CSeq: 1 INVITE > > User-Agent: CS2000_NGSS/8.0 > > Remote-Party-ID: <sip:0289089500@10.39.1.16;user=phone>; party=calling; > privacy=off; screen=yes > > Allow: ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK > > P-Asserted-Identity: <sip:0289089500@10.39.1.16;user=phone> > > Max-Forwards: 69 > > Contact: <sip:77.239.128.10:5060;transport=UDP> > > Supported: 100rel > > Content-Type: application/sdp > > Content-Length: 236 > > > > v=0 > > o=PVG 1305215180830 1305215180830 IN IP4 77.239.128.10 > > s=- > > p=+1 6135555555 > > c=IN IP4 77.239.128.10 > > t=0 0 > > m=audio 49264 RTP/AVP 18 8 0 13 101 > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-15 > > a=ptime:20 > > a=fmtp:18 annexb=no > > > > > > > > SIP/2.0 403 Proxy forbidden > > Via: SIP/2.0/UDP > 77.239.128.10:5060;branch=z9hG4bKf89p48m8dkuco1tumbnfaemc44;rport=5060;received=77.239.128.10 > > From: > <sip:0289089500@10.39.1.16:5060;user=phone>;tag=SDml08d01--45026-1f4d3a5-203bbf09-1f4d3a5 > > To: > <sip:0698962030@10.39.6.207:5060;user=phone>;tag=4471d251144740103ef2d5e5631bdc66.799c > > Call-ID: SDml08d01-daae53b1c06b7367e53d68ad67a63d13-ao92061 > > CSeq: 1 INVITE > > Server: Kamailio (1.4.4-notls (x86_64/linux)) > > Content-Length: 0 > > > > > > > > > > > > > > > > > > > > > > ACK sip:0698962030@79.12.254.153:5060;user=phone SIP/2.0 > > Via: SIP/2.0/UDP > 77.239.128.10:5060;branch=z9hG4bKf89p48m8dkuco1tumbnfaemc44;rport=5060;received=77.239.128.10 > > From: > <sip:0289089500@10.39.1.16:5060;user=phone>;tag=SDml08d01--45026-1f4d3a5-203bbf09-1f4d3a5 > > To: > <sip:0698962030@10.39.6.207:5060;user=phone>;tag=4471d251144740103ef2d5e5631bdc66.799c > > Call-ID: SDml08d01-daae53b1c06b7367e53d68ad67a63d13-ao92061 > > CSeq: 1 ACK > > Max-Forwards: 69 > > Content-Length: 0 > > > > > > > INVITE sip:0698962030@79.12.254.153:5060;user=phone SIP/2.0 > > Via: SIP/2.0/UDP 77.239.128.10:5060;branch=z9hG4bK4t5g11q75ccl8k5vnbgabnlct2 > > From: > <sip:026683011@10.39.1.16:5060;user=phone>;tag=SDa7sn401--45026-1f4d3ad-69943699-1f4d3ad > > To: <sip:0698962030@10.39.6.207:5060;user=phone> > > Call-ID: SDa7sn401-c505e58e02da5e4449475a8d28dbcb07-ao92061 > > CSeq: 1 INVITE > > User-Agent: CS2000_NGSS/8.0 > > Remote-Party-ID: <sip:026683011@10.39.1.16;user=phone>; party=calling; > privacy=off; screen=yes > > Allow: ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK > > P-Asserted-Identity: <sip:026683011@10.39.1.16;user=phone> > > Max-Forwards: 69 > > Contact: <sip:77.239.128.10:5060;transport=UDP> > > Supported: 100rel > > Content-Type: application/sdp > > Content-Length: 236 > > > > v=0 > > o=PVG 1305215458700 1305215458700 IN IP4 77.239.128.10 > > s=- > > p=+1 6135555555 > > c=IN IP4 77.239.128.10 > > t=0 0 > > m=audio 41892 RTP/AVP 18 8 0 13 101 > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-15 > > a=ptime:20 > > a=fmtp:18 annexb=no > > > > > > > > SIP/2.0 403 Proxy forbidden > > Via: SIP/2.0/UDP > 77.239.128.10:5060;branch=z9hG4bK4t5g11q75ccl8k5vnbgabnlct2;rport=5060;received=77.239.128.10 > > From: > <sip:026683011@10.39.1.16:5060;user=phone>;tag=SDa7sn401--45026-1f4d3ad-69943699-1f4d3ad > > To: > <sip:0698962030@10.39.6.207:5060;user=phone>;tag=4471d251144740103ef2d5e5631bdc66.fe6e > > Call-ID: SDa7sn401-c505e58e02da5e4449475a8d28dbcb07-ao92061 > > CSeq: 1 INVITE > > Server: Kamailio (1.4.4-notls (x86_64/linux)) > > Content-Length: 0 > > > > > > > > > > > > > > > > > > > > > > > > sip:0698962030@79.12.254.153:5060;user=phone SIP/2.0 > > Via: SIP/2.0/UDP > 77.239.128.10:5060;branch=z9hG4bK4t5g11q75ccl8k5vnbgabnlct2;rport=5060;received=77.239.128.10 > > From: > <sip:026683011@10.39.1.16:5060;user=phone>;tag=SDa7sn401--45026-1f4d3ad-69943699-1f4d3ad > > To: > <sip:0698962030@10.39.6.207:5060;user=phone>;tag=4471d251144740103ef2d5e5631bdc66.fe6e > > Call-ID: SDa7sn401-c505e58e02da5e4449475a8d28dbcb07-ao92061 > > CSeq: 1 ACK > > Max-Forwards: 69 > > Content-Length: 0 > > > > > > > > > > _______________________________________________ > 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

3 4

SEMS 1.4.0 release
by Stefan Sayer 24 May '11

24 May '11

We are happy to announce the availability of the SIP Express Media Server version 1.4.0. SEMS is a high performance media and application server for SIP based VoIP networks. This release features a powerful Session Border Controller (SBC) module. From completely transparent B2BUA to customized URI/From/To, strictly filtered (messages, headers, codecs) with RTP anchoring, Session Timer enforcement, prepaid and call timer, the SBC facilitates interconnect and core routing in a simple and secure way. Thanks to the new multihoming support, SEMS can now be employed at the border of the networks. This addition also allows to overcome the bottleneck of one NIC - giving the possibility to fully exploit SEMS' great performance. In the app development area, the DSM language has matured to become a viable candidate also for implementing complex application logic, thanks to language constructs like for, if and functions. SEMS can be downloaded in source from its ftp site at: ftp.iptel.org/pub/sems/1.4/1.4.0/ Packages for debian/ubuntu should appear shortly on the OBS page: https://build.opensuse.org/project/show?project=home:team-sems Contributors to this release: Stefan Sayer, Raphael Coeffic, Bogdan Pintea, Matthew Williams, Peter Lemenkov, Anton Zagorskiy, Robert Szokovacs, Jon Bonilla, Thomas Gelf, Erik Dekkers, Tobias Lindgren, David J Many thanks to the companies sponsoring SEMS development, especially FRAFOS GmbH and TelTech Systems Inc. "E stands for Express." * Apologies if you receive multiple copies of this message. *

4 5

~ operator has no effect
by Juha Heinanen 24 May '11

24 May '11

according to core cookbook: ~ : bitwise NOT however, looks like it has no effect. i have $var(test) = ~2; xlog("L_INFO", "test is <$var(test)>\n"); and i get to syslog: May 17 18:10:40 sip /usr/sbin/sip-proxy[29280]: INFO: test is <2> have i misunderstood the operator or is there a bug? -- juha

3 5

Re: [SR-Users] IPv6
by Daniel-Constantin Mierla 24 May '11

24 May '11

On 5/20/11 6:33 PM, Jon Farmer wrote: > On 20 May 2011 17:25, Daniel-Constantin Mierla<miconda(a)gmail.com> wrote: > >> the module is good, but not for this case -- the warning seems to be from >> startup, saying that no reverse dns could be done to the ipv6 listen address >> and it is harmless (done for the purpose of building aliases list). >> >> Do you get the warning at runtime? >> >> Normally, the ipv6 should be in between square brackets. >> > Adding the address in question to my hosts file temporarily takes the > warning away :) ok, let me know how is going with the testing of ipv6 -- the support is in core since 2002, but not many entered the challenge so far :-) Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

4 4

newbie INVITE routing logic help
by Skyler 24 May '11

24 May '11

Hi all, Need some newbie advice on how to write INVITE routing for my scenario. I currently have kamailio+asterisk-realtime integration working in the following way: Outbound calls: USER-A -> Kamailio(3.1.3) -> Asterisk(1.6.2.18) ;if dest local then Asterisk(1.6.2.18) -> Kamailio(3.1.3) -> USER-B ;if dest not local then Asterisk(1.6.2.18) -> SIP/PSTN This works, though I'd prefer to use Kamailio + LCR and gateway failover as Asterisk doesn't do this well. Here's what I want to do: (for now, small steps) Outbound Calls: USER-A -> Kamailio(3.1.3) -> Asterisk(1.6.2.18) ;if dest local then Asterisk(1.6.2.18) -> Kamailio(3.1.3) -> USER-B ;if dest not local then Asterisk(1.6.2.18) -> Kamailio(3.1.3)+(lcr) -> SIP/PSTN I know I have to set kamailio as a GW in asterisk instead of the PSTN GW's and add the PSTN GW's to kamailio instead, though I'm at a loss as to how to write the routing logic so that kamailio sends calls from asterisk to PSTN. route[LOCATION] { ...non-INVITE check ends here... if(route(FROMASTERISK)) { # coming from Asterisk --- Does this mean check kamailio local users first before continuing? if (!lookup("location")) { switch ($rc) { case -1: case -3: t_newtran(); t_reply("404", "Not Found"); exit; case -2: sl_send_reply("405", "Method Not Allowed"); exit; } } ## -- If match 11 digits and starts with '1' or if match 10 digits and starts with [2-9] if($rU=~"^(1\d{10})$") || if($rU=~"^([2-9]\d{9})$") { xlog("L_INFO", " FROMASTERISK: This will become route(NORTHAMERICA); \n"); exit; ## to be removed } else { xlog("L_INFO", " FROMASTERISK: This will become route(INTERNATIONAL); \n"); exit; ## to be removed } # new call - send to Asterisk route(TOASTERISK); } Will this logic work? I haven't determined how to write route(NORTHAMERICA) or route(INTERNATIONAL) yet so I'm just logging to debug the logic and syntax. Thanks in advance.

1 0

Kamailio to handle call states
by Yeris Antonio Madrigal Castro 23 May '11

23 May '11

Hi I am planing to change my old T1-E1 pbx, wich is AVAYA, for an asterisk cluster. I am planning this for a call center and I spec 80 currents calls for 30 agents sometimes, so I need to handle big queues and call and agent states, is it possible for kamailio handle the call states and agent states, and leave asterisk for the call center functions? Thanks in advance

1 0

TLS - Kamailio Asterisk - Registration issue
by Lucas Alvarez 23 May '11

23 May '11

Hi, I have confgured Kamailio with TLS in an Kamailio-Asterisk implementation. The phones (Bria - Eyebeam - Aastra 57i) are registering in Kamailio without problems, but the registration is not being forwarded to Asterisk. Kamailio is listening on port tls:5061 only and Asterisk on udp:5080. Should Kamailio has to listen on port 5060 also for UAC module functions? If I enable the port 5060, how do I prevent common SIP - UDP registrations? I mean, I want the registrations to be over TLS only. I would appreciate if someone can put me on the right path. Thanks in advance. Lucas Alvarez

2 1

change ptime problem
by hala alramli 22 May '11

22 May '11

i am trying to route to pstn provider,but i faced a problem . my provider accept ptime 20 while i am send the call with ptime 30 how i can change ptime i have kamailio with freeswitch any help please

4 4

Kamailio + Asterisk 1.6.2 Realtime - REGFWD : 401 not authorized
by Skyler 22 May '11

22 May '11

Hi all, After 3 hours stuck on this I have to ask the group. I am setting up Kam 3.1.3 +Ast 1.6.2.18 + realtime following Daniel's guide on the Asipto site. The problem I see is 401 not authorized when uac tries to register. Below is ngrep . do you see something I am missing or have any pointers? I can't seem to get any devices to register with asterisk. --- TIA interface: any filter: (ip or ip6) and ( port 5060 ) U 2011/05/21 23:40:39.333181 192.168.1.132:5060 -> 192.168.1.104:5060 REGISTER sip:192.168.1.104 SIP/2.0. Via: SIP/2.0/UDP 192.168.1.132:5060;branch=z9hG4bK-cd60fa71. From: testing <sip:4444@192.168.1.104>;tag=16b905e66a42787eo0. To: testing <sip:4444@192.168.1.104>. Call-ID: 5407498c-e755970e(a)192.168.1.132. CSeq: 55278 REGISTER. Max-Forwards: 70. Authorization: Digest username="4444",realm="192.168.1.104",nonce="Tdiwvk3Yr5JAcX9Iljh/AX0BieRt73Y v",uri="sip:192.168.1.104",algorithm=MD5,response="68bc7ad2de02caa48b8cbb22b 6de7aa6". Contact: testing <sip:4444@192.168.1.132:5060>;expires=30. User-Agent: Linksys/PAP2-3.1.23(LS). Content-Length: 0. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. Supported: x-sipura, replaces. U 2011/05/21 23:40:39.347248 192.168.1.104:5060 -> 192.168.1.132:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.1.132:5060;branch=z9hG4bK-cd60fa71;rport=5060. From: testing <sip:4444@192.168.1.104>;tag=16b905e66a42787eo0. To: testing <sip:4444@192.168.1.104>;tag=d0c53e97c845e2d86a9876bb195e3450.e92e. Call-ID: 5407498c-e755970e(a)192.168.1.132. CSeq: 55278 REGISTER. Contact: <sip:4444@192.168.1.132:5060>;expires=60;received="sip:192.168.1.132:5060". Server: kamailio (3.1.3 (i386/linux)). Content-Length: 0. U 2011/05/21 23:40:39.349726 192.168.1.104:5060 -> 192.168.1.104:5080 REGISTER sip:192.168.1.104:5080 SIP/2.0. Via: SIP/2.0/UDP 192.168.1.104;branch=z9hG4bK0f62.635b95f3.0. To: sip:4444@192.168.1.104. From: sip:4444@192.168.1.104;tag=a707bfce77c9367d1734afac7b3de6ca-f4de. CSeq: 10 REGISTER. Call-ID: 7f1ddb01-6717(a)192.168.1.104. Content-Length: 0. User-Agent: kamailio (3.1.3 (i386/linux)). Contact: <sip:4444@192.168.1.104:5060>. Expires: 30. U 2011/05/21 23:40:39.351790 192.168.1.104:5080 -> 192.168.1.104:5060 SIP/2.0 401 Unauthorized. v: SIP/2.0/UDP 192.168.1.104;branch=z9hG4bK0f62.635b95f3.0;received=192.168.1.104. f: sip:4444@192.168.1.104;tag=a707bfce77c9367d1734afac7b3de6ca-f4de. t: sip:4444@192.168.1.104;tag=as64ca9fe3. i: 7f1ddb01-6717(a)192.168.1.104. CSeq: 10 REGISTER. Server: taridium ipbx. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. k: replaces, timer. WWW-Authenticate: Digest algorithm=MD5, realm="192.168.1.104", nonce="5bad77f5". l: 0. U 2011/05/21 23:40:53.322348 192.168.1.104:5060 -> 192.168.1.132:5060 .... U 2011/05/21 23:40:57.180694 192.168.1.132:5060 -> 192.168.1.104:5060 NOTIFY sip:192.168.1.104 SIP/2.0. Via: SIP/2.0/UDP 192.168.1.132:5060;branch=z9hG4bK-56c97474. From: testing <sip:4444@192.168.1.104>;tag=16b905e66a42787eo0. To: <sip:192.168.1.104>. Call-ID: dbc8d986-9fbb95e(a)192.168.1.132. CSeq: 47 NOTIFY. Max-Forwards: 70. Event: keep-alive. User-Agent: Linksys/PAP2-3.1.23(LS). Content-Length: 0. U 2011/05/21 23:40:57.187840 192.168.1.104:5060 -> 192.168.1.132:5060 SIP/2.0 200 OK - keepalive. Via: SIP/2.0/UDP 192.168.1.132:5060;branch=z9hG4bK-56c97474;rport=5060. From: testing <sip:4444@192.168.1.104>;tag=16b905e66a42787eo0. To: <sip:192.168.1.104>;tag=d0c53e97c845e2d86a9876bb195e3450.e05a. Call-ID: dbc8d986-9fbb95e(a)192.168.1.132. CSeq: 47 NOTIFY. Server: kamailio (3.1.3 (i386/linux)). Content-Length: 0.

2 2

how to add RR with advertised_address in front of normal RR?
by Andrew Pogrebennyk 22 May '11

22 May '11

I've been playing with sip load-balancer in EC2 but so far haven't been able to figure out how to insert Record-Route header with advertised_address. AFAIK kamailio rr module adds 2 RR headers (one with the inbound address and one with the outbound address) when your transaction comes with one interface/port and leaves via another. In EC2 the receive socket is the same as send socket and I've had no luck trying to get kamailio insert advertised address as inbound RR and then the outbound RR, by calling record_route_preset() several times in a row too. This is a problem if lb is in front of another proxy. As I see this question has been raised on the mailing lists as far as 6 years ago - I would be surprised if no one has come up with a solution yet. So how to do record-routing in a draft-ietf-sip-record-route-fix AKA rfc5658 style on the machine with one physical interface? -- Sincerely, Andrew Pogrebennyk

3 2

Forwarding client IP to secondary SIP server (asterisk)
by Muhammad Naseer Bhatti 22 May '11

22 May '11

Hello, I am testing Kamailio (OpenSER) as a frontend forwarding calls to asterisk. The setup is working fine and I am able to make call fine. However, Asterisk, naturally, sees calls/connections coming from Kamailio (OpenSER). Is there a way I can forward the client (calle to Kamailio (OpenSER)) IP to asterisk so asterisk can authenticate it? -B

2 1

syntax error?
by Skyler 22 May '11

22 May '11

Hi all, I am attempting to handle NOTIFY within my script but there is a definite error in my syntax. I believe it has to do with $hdr(Event). Anyone know how I should re-write this? if (is_method("NOTIFY") && is_domain_local("$rd") && $hdr(Event) =~ "keep-alive" ) { sl_send_reply("200", "OK - keep-alive"); exit; } TIA

2 2

Re: [SR-Users] syntax error?
by Skyler 22 May '11

22 May '11

Found it.will google more from now on. Sorry. if ( is_method("NOTIFY") && uri==myself && $hdr(Event) =~ "keep-alive" ) { sl_send_reply("200", "OK - keepalive"); xlog("L_INFO", "$hdr(Event) detected - ok sent\n"); exit; } From: Skyler [mailto:skchopperguy@gmail.com] Sent: Saturday, May 21, 2011 7:40 PM To: 'sr-users(a)lists.sip-router.org' Subject: syntax error? Hi all, I am attempting to handle NOTIFY within my script but there is a definite error in my syntax. I believe it has to do with $hdr(Event). Anyone know how I should re-write this? if (is_method("NOTIFY") && is_domain_local("$rd") && $hdr(Event) =~ "keep-alive" ) { sl_send_reply("200", "OK - keep-alive"); exit; } TIA

1 0

Re: [SR-Users] Modules directory missing?
by Skyler 22 May '11

22 May '11

Found it. /usr/lib/kamailio/modules_k/:/usr/lib/kamailio/modules/ From: Skyler [mailto:skchopperguy@gmail.com] Sent: Saturday, May 21, 2011 3:29 PM To: 'sr-users(a)lists.sip-router.org' Subject: Modules directory missing? Hello, I am new to Kamailio and have installed via rpm repository on CentOS 5.5 Where is the modules directory located? I cannot load any modules and Kamailio will not start. I've checked these locations but they don't exist. Is there another package I need to install or where did I go wrong? /usr/local/lib/kamailio/modules_k/ /usr/local/lib/kamailio/modules/

1 0

Modules directory missing?
by Skyler 21 May '11

21 May '11

Hello, I am new to Kamailio and have installed via rpm repository on CentOS 5.5 Where is the modules directory located? I cannot load any modules and Kamailio will not start. I've checked these locations but they don't exist. Is there another package I need to install or where did I go wrong? /usr/local/lib/kamailio/modules_k/ /usr/local/lib/kamailio/modules/

1 0

Re: [SR-Users] ACK to 486 is not being proxied
by Carl Wagner 21 May '11

21 May '11

Daniel, Sorry, I had meant to send it to the list. You are correct about missing messages. The ACK is being sent, but it is sent to it's self (loop condition) over the 'lo' device. So basically its a design flaw on my part of the kamailio.cfg file. My current config works for initial messages from the provider and from subscribers. But for initial messages from the Asterisk boxes, I have Asterisk send to the Kamailio box and then have Kamailio look at the message and then route it with a forward(x.x.x.x). While that works for successful calls it is what is breaking the ACK to the 486, and I think it is also breaking Cancel. Do you know of a good example config file where: 1) SIP message from provider hits Kamailio and is routed to one of several Asterisk boxes (load balancing, using ds_select) 2) Messages from a subscriber will be routed to one of the Asterisk boxes (load balancing, using ds_select) 3) Messages from an Asterisk box will be routed to the provider OR a subscribers SIP device based on a fixed string in the R-URI Or you can laugh at what I have: (constructive criticism is welcome from anyone!) (I had to trim this message as it was exceeding the 40K limit) #!KAMAILIO # ####### Defined Values ######### # - 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 ####### Global Parameters ######### # debug=4 # log_stderror=yes debug=3 log_stderror=no memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=4 /* 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:2.2.2.164:5060 /* port to listen to * - can be specified more than once if needed to listen on many ports */ port=5060 dns=no rev_dns=no open_files_limit=4096 disable_core_dump=yes disable_tcp=yes check_via=0 reply_to_via=0 # we may be able to use the above with force_rport ####### Modules Section ######## # set paths to location of modules mpath="/usr/lib/kamailio/modules_k/:/usr/lib/kamailio/modules/" loadmodule "db_mysql.so" loadmodule "mi_fifo.so" loadmodule "kex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" # loadmodule "ctl.so" loadmodule "mi_rpc.so" loadmodule "acc.so" loadmodule "auth.so" loadmodule "auth_db.so" loadmodule "dispatcher.so" loadmodule "avpops.so" # ----------------- 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) modparam("tm", "fr_timer", 30000) 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) # ----- 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) # the line below puts extra stuf at the end of the log line modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ # ----- usrloc params ----- /* enable DB persistency for location entries */ # appears to use table 'location' and this table has data in it modparam("usrloc", "db_url", "mysql://kamailio:password@my.db.com/kamailio") modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", 1) # ----- auth_db params ----- # appears to use table 'subscriber' and this table has data in it modparam("auth_db", "db_url", "mysql://kamailio:password@my.db.com/kamailio") modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", 1) modparam("dispatcher", "list_file", "/etc/kamailio/dispatcher.list") modparam("dispatcher", "force_dst", 1) modparam("dispatcher", "flags", 3) modparam("avpops","db_url","mysql://kamailio:password@my.db.com/AppServer") modparam("avpops","avp_table","User") ####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route route { # initial checks (check for bad/malformed/unsupported messages) route(REQINIT); # if it is not a OPTIONS message - log it if (!is_method("OPTIONS")) { xlog("L_INFO", "======== processing new message\n"); xlog("L_INFO", "MAIN: SIP Request: [$rm] ruri=[$ru] (from [$fu] to [$tu], Call-ID=[$ci]) CSeq=[$cs]\n"); } # handle requests within SIP dialogs (active ongoing call - has TO: tag) does not return route(WITHINDLG); ### only initial requests (no To: tag) from here on # CANCEL processing if (is_method("CANCEL")) { # - this needs some work and cancels are not being replied to or forwared # cancle is a hop-by-hop message (send a 200 OK then forward it on) # t_check_trans = if cancel, this is supposed to be true if the Invite transaction exits # see: http://www.kamailio.org/docs/modules/1.1.x/tm.html#AEN460 if (t_check_trans()) { xlog("L_INFO", "MAIN: Recieved Cancel, t_check_trans is true\n"); t_relay(); exit; } xlog("L_INFO", "MAIN: Recieved Cancel, t_check_trans is FALSE\n"); # forward it on (not sure why we are getting here with valid calls that need to be canceled) # this may be bad as somone might be able to "cancel storm" us?? t_relay(); exit; } # this was added to handle OPTIONS messages route(HANDLEOPTIONS); # I 'think' this absorbs retransmissions of Invites t_check_trans(); if (!is_method("REGISTER")) { if ($rU==$null) { # request with no Username in RURI xlog("L_INFO", "Error: main route: request with no Username in RURI\n"); sl_send_reply("484","Address Incomplete"); exit; } } # set up a trigger for failures of INVITEs if (is_method("INVITE")) { xlog("L_INFO", "main_loop: SIP Request: ruri=[$ru] (method [$rm] from [$fu] to [$tu])\n"); t_on_failure("FAIL_ONE"); } # at this point we need to determine how to route the call # if the call is from PROVIDER, it has to go to an AST box (L3 should never go directly to a subscriber) # if the call is from a subscriber, it has to go to an AST box (sub can't dial directly out) # if the call is from an AST box we need to determine if it goes to L3 or a subscriber # if to a subscriber the ToURI should contain "SIP_" # log all invites to syslog # if (is_method=="INVITE")) # { # xlog("L_INFO", "main_loop: SIP Request: ruri=[$ru] (method [$rm] from [$fu] to [$tu])\n"); # }; # handle the easy cases first # from PROVIDER if ($src_ip=="3.3.3.40") { route(FROMPROVIDER); exit; # just to be sure } # if from an AST box (will determine wether to send it to PROVIDER or a subscriber) if ($src_ip=~"2\.2\.2\.1") # regexp match { route(FROMAST); exit; # just to be sure } force_rport(); # attemp to make kamailio use the emhemeral port of the sender #=============== Only subscribers left from here down # handle registrations (from a sip client) route(REGISTER); # authentication for everyone route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } route(FROMSUBSCRIBER); # route(RELAY); xlog("L_INFO", "ERROR - no route chosen \n"); } ################################################################### # Per SIP request initial checks route[REQINIT] { # xlog("L_INFO", " REQINIT: Initial Checks\n"); if (!mf_process_maxfwd_header("10")) { xlog("L_WARN", " REQINIT: Too Many Hops (Request: [$rm])\n"); sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("L_WARN", " REQINIT: Malformed SIP message from $si:$sp (Request: [$rm]) \n"); exit; } # xlog("L_INFO", " leaving [REQINIT] \n"); # unsupported messages here if (is_method("SUBSCRIBE")) { # probably want to comment the next line out after it is working xlog("L_WARN", "REQINIT: Subscribe not supported: SIP Request: [$rm] ruri=[$ru] (from [$fu] to [$tu], Call-ID=[$ci]) CSeq=[$cs]\n"); exit; } # if no errors, just fall through and return to caller } ################################################################### # Handle requests within SIP dialogs (request has a TO: Tag) route[WITHINDLG] { # xlog("L_INFO", " WITHINDLG: starting up (WithInDialog)\n"); if (has_totag()) { xlog("L_INFO", " WITHINDLG: SIP Request: [$rm] ruri=[$ru] (from [$fu] to [$tu], Call-ID=[$ci], CSeq=[$cs])\n"); xlog("L_INFO", " WITHINDLG: has TO: tag\n"); # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { xlog("L_INFO", " WITHINDLG: has TO: tag AND loose_route is true\n"); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } xlog("L_INFO", " WITHINDLG: sending to relay\n"); route(RELAY); } else # not loose_route { xlog("L_INFO", " WITHINDLG: has TO: tag AND louse_route is NOT true\n"); if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { xlog("L_INFO", " WITHINDLG: has TO: tag AND loose_route is NOT true and is_method = ACK\n"); if ( t_check_trans() ) # see if a message is related to a transaction { # Reference: http://www.kamailio.org/docs/modules/stable/modules/tm.html#t_check_trans xlog("L_INFO", " WITHINDLG: has TO: tag AND loose_route is NOT true and is_method = ACK and t_check_trans=true\n"); # 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 xlog("L_INFO", " WITHINDLG: has TO: tag AND loose_route is NOT true and is_method = ACK and t_check_trans=FALSE\n"); xlog("L_INFO", " WITHINDLG: the ACK to a 486 was not being processed so I am adding t_relay here\n"); # $var(a) = t_relay(); # $var(a) = forward(); t_relay(); xlog ("L_INFO", " WITHINDLG: (ReturnCode = [$var(a)] exiting)\n"); exit; } } # not an ACK - as both branches aove exit xlog("L_INFO", " WITHINDLG: has TO: tag AND loose_route is NOT true and is_method not = ACK \n"); sl_send_reply("404","Not here"); } exit; } # xlog("L_INFO", " leaving [WITHINDLG] (WithInDialog)\n"); # if msg has to: tag it should be processed above and then exit, else just return for further processing } ################################################################### # By CEW 2Feb2011 route[REGISTER] { if (is_method("REGISTER")) { # NOTE: need to review: http://www.kamailio.org/docs/scripting.html on how they handle registration # at this point all that should be left are sip clients - they are coming over the open internet so authenticate them! # xlog("L_INFO", " REGISTER: processing register from sip client, fu=$fu \n"); # authenticate the REGISTER requests (uncomment to enable auth) # www_authenticate will send a 401 Unauthorized (and will contain WWW-Authenticate in the reply) # proxy_authenticate will send a 407 Proxy Authentication Required. (and will contain Proxy-Authenticate in the reply) # uses the kamailio->subsriber DB table # if (!proxy_authenticate("2.2.2.164", "subscriber")) if (!www_authenticate("2.2.2.164", "subscriber")) { $var(a) = $rc; # save the return from the www_authenticate # -1 = generic error # -2 = invalid password # -3 = invalid user ( or domain is wrong) if ($var(a) == -2) { xlog ("L_INFO", " REGISTER ERROR: bad password, (ReturnCode = $var(a) exiting)\n"); exit; } if ($var(a) == -3) { xlog ("L_INFO", " REGISTER ERROR: invalid user, (ReturnCode = $var(a) exiting)\n"); exit; } xlog("L_INFO", " REGISTER: issuing a challenge, 401 (and then exit script)\n"); # send a "401 Unauthorized", with realm=2.2.2.164, caller should resend invite with auth info www_challenge("2.2.2.164", "1"); # send a 401 # proxy_challenge("2.2.2.164", "1"); # send a 407 # xlog("L_INFO", " REGISTER: exiting\n"); exit; } xlog("L_INFO", " REGISTER: Match found in database for register\n"); # make sure the Authorization: username matches the To: username # $au = user part of username from Authorization: or Proxy-Authorization header (from the resent Invite, after the challenge) # $tU = reference to username in URI of 'To' header if ($au!=$tU) { xlog("L_INFO", " REGISTER: sending 403 Forbidden auth ID\n"); sl_send_reply("403","Forbidden auth ID"); exit; } # now store it in the DB and then exit if (!save("location")) { xlog("L_INFO", " REGISTER: Error saving location, sending SIP error to other end\n"); sl_reply_error(); } exit; } } ################################################################### # Authentication route route[AUTH] { xlog("L_INFO", " entering route[AUTH] Authentication route\n"); # authenticate if from local subscriber # I think this should be true for LOCAL sip callers (in production there shouldn't be any of these) if (from_uri==myself) { xlog("L_INFO", " AUTH: from_uri = myself ($fu)\n"); if (!proxy_authenticate("$fd", "subscriber")) { $var(a) = $rc; # save the return from the www_authenticate # -1 = generic error - like no valid auth # -2 = invalid password # -3 = invalid user (or domain is wrong) if ($var(a) <= -2) { xlog ("L_INFO", " AUTH: ERROR: Location lookup, ReturnCode = $var(a) exiting\n"); exit; } # xlog("L_INFO", " AUTH: no authenticate credentials supplied, sending challenge (407), from_uri=$fu \n"); xlog("L_INFO", " AUTH: no authenticate credentials supplied, sending challenge (401), from_uri=$fu \n"); # send the proxy challenge and exit. Let the caller send another invite with authentication info. proxy_challenge("$fd", "0"); exit; } xlog("L_INFO", " AUTH: proxy_autorization OK! \n"); # make sure the Authorization: username matches the To: username # $au = user part of username from Authorization: or Proxy-Authorization header (from the resent Invite, after the challenge) # $fU = reference to username in URI of 'From' header if ($au!=$fU) { xlog("L_INFO", " AUTH: Forbidden, sending 403\n"); sl_send_reply("403","Forbidden auth ID"); exit; } xlog("L_INFO", " AUTH: Authorization accepted! Consuming credentials\n"); consume_credentials(); # caller authenticated } else { # !! NOTE: this branch has not been tested as I cant simulate it in my lab!!! # a lot of it was just made up so it does need a lot of testing. # subscriber from the Internet # this is where all our sip clients should hit xlog("L_INFO", " AUTH: EXTERNAL subscriber, need to authenticate from_uri $fu\n"); # caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (!uri==myself) { xlog("L_INFO", " AUTH: NOT RELAYING!, (uri!=myself) sending 403\n"); sl_send_reply("403","Not relaying"); exit; } # not sure if we need to do a proxy_authenticate or www_authenticate here - need to test in production if (!proxy_authenticate("$fd", "subscriber")) { xlog("L_INFO", " AUTH: sending proxy_challenge, from_uri=$fu, exiting \n"); # send the proxy challenge and exit. Let the caller send another invite with authentication info. proxy_challenge("$fd", "0"); xlog("L_INFO", " AUTH: sent proxy_challenge, exiting\n"); exit; } # make sure the Authorization: username matches the To: username # $au = user part of username from Authorization: or Proxy-Authorization header (from the resent Invite, after the challenge) # $fU = reference to username in URI of 'From' header if ($au!=$fU) { xlog("L_INFO", " AUTH: Forbidden auth ID, from_uri=$fu, exiting \n"); sl_send_reply("403","Forbidden auth ID"); exit; } xlog("L_INFO", " AUTH: Consuming credentials\n"); consume_credentials(); } xlog("L_INFO", " leaving route[AUTH] Authentication route\n"); return; } ################################################################### # if the call is from PROVIDER, it has to go to an AST box (L3 should never go directly to a subscriber) # if (uri=="3.3.3.40") route[FROMPROVIDER] { # xlog("L_INFO", " FROMPROVIDER: starting up \n"); if (is_method("REGISTER")) { xlog("L_INFO", " FROMPROVIDER: REGISTER from PROVIDER, sending 200 OK fd=$fd \n"); sl_send_reply("200", "ok"); exit; }; # 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 }; # if a conference call, send it to ast1 # ds_select_dst(set, alg) the method selects a destination from address set # set - first column from dest file (dispacher.list) # alg 0 = hash over callid if (uri=~"12223334444") { xlog("L_INFO", " FROMPROVIDER: Conference call - sending to ast1 \n"); ds_select_dst("3", "0"); forward(); exit; }; # avp_check(name, op_value) check the value of the avp against an operator and value # op_value = operator '/' value ['/'flags] # $fu = # $ru = SIP Request's URI # if the call came from ast0 (2.2.2.167) via PROVIDER route it to ast1 or we will have a loop condition # $ci = the call ID # AVP = attribute-value-pair if (avp_check("$ci","re/.*(a)2.2.2.167.*/g")) { xlog ("L_INFO", " FROMPROVIDER: Call originating on ast0, sending call to ast1 [$ru]\n"); ds_select_dst("2", "0"); forward(); exit; }; # if the call came from ast1 (2.2.2.168) via PROVIDER route it to ast0 or we will have a loop condition if (avp_check("$ci","re/.*(a)2.2.2.168.*/g")) { xlog ("L_INFO", " FROMPROVIDER: Call originating on ast1, sending call to ast0 [$ru]\n"); ds_select_dst("3", "0"); forward(); exit; }; # Extract the telephone number from the To-URI $avp(i:999)=$tU; # insted of the next line, should we just do a regexp to remove all non-numeric? avp_subst("$avp(i:999)","/\+//g"); # remove the literal string "+" from the variable, if any # xlog("L_INFO", " FROMPROVIDER: about to do \"SELECT AppName FROM Application WHERE TelephoneNumber = $avp(i:999) \" \n"); avp_db_query("SELECT AppName FROM Application WHERE TelephoneNumber = '$avp(i:999)'","$avp(i:888)"); # xlog("L_INFO", " FROMPROVIDER: Query result =[$avp(i:888)] \n"); # see if we got anything back from the DB, if we did it would look like "product" or some future varient if(!is_avp_set("$avp(i:888)")) # if avp 888 is not set, dump the call { xlog ("L_INFO", " FROMPROVIDER: Bad number. Sending 404. method=[$rm] ruri=[$ru] to=[$tu] from=[$fu]\n"); # I don't think the next line is needed as it appears this table is never read (just written to) avp_db_query("INSERT IGNORE INTO InvalidTN VALUES ('$avp(i:999)')"); sl_send_reply("404", "Not Found"); # do we really want to do this? Or shoud we just silently drop it? exit; }; # this is where we do the main load balancing, ast0 or ast1 xlog ("L_INFO", " FROMPROVIDER: Normal Round-Robin call, routing to [$ru], from [$fu]\n"); # ds_select_dst("1", "0"); ds_select_dst("3", "0"); forward(); # Note: forward does return, so we could do post call processing here exit; } ################################################################### # if the call is from an AST box we need to determine if it goes to L3 or a subscriber # if to a subscriber the ToURI should contain "SIP_" # if (uri=~"2\.2\.2\.1") from our ast box route[FROMAST] { xlog("L_INFO", " FROMAST: starting up \n"); if (is_method("REGISTER")) { xlog("L_INFO", " FROMAST: REGISTER from AST, sending 200 OK fd=$fd \n"); sl_send_reply("200", "ok"); exit; }; # 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 } # this block is probalby not needed, except possilby for logging $avp(i:999)=$tU; # Extract the telephone number from the To-URI avp_subst("$avp(i:999)","/\+//g"); # remove the literal string "sip:+" from the variable xlog("L_INFO", " FROMAST: avp(999) = $avp(i:999)\n"); # at this point we need to see if the number contains a "SIP_". If so route it to the subscriber, else route it to L3 xlog("L_INFO", " FROMAST: tU=$tU\n"); if ($tU=~"SIP_") { xlog("L_INFO", " FROMAST: contains \"SIP_\" so routing to subscriber\n"); route(TOSUBSCRIBER); exit; # should be handled in TOSUBSCRIBER, but just to make sure we bail here and don't continue } xlog("L_INFO", " FROMAST: does not contain SIP_ so routing to PROVIDER\n"); # need to rewrite the RURI and To URI $td="3.3.3.40:5060"; $rd="3.3.3.40:5060"; forward("3.3.3.40"); # send it on to PROVIDER } ################################################################### # if the call is from a subscriber, it has to go to an AST box (subscribers can't dial directly out) route[FROMSUBSCRIBER] { xlog("L_INFO", " FROMSUBSCRIBER: starting up \n"); # at this point we should have already been authenticated by route[AUTH] so just route it to an ast box # this is where we do the main load balancing, ast0 or ast1 (round robin) ds_select_dst("3", "0"); forward(); xlog("L_INFO", " leaving route[FROMSUBSCRIBER] \n"); exit; # just in case } ################################################################### # The call needs to be routed to a scubscriber route[TOSUBSCRIBER] { xlog("L_INFO", " TOSUBSCRIBER: starting up \n"); # this is NOT tested yet! # do a DB lookup on kamailio->location to see where to route (should only be for SIP-to-Subscribers) if (!lookup("location")) { switch ($rc) { case -1: case -3: t_newtran(); xlog("L_INFO", " TOSUBSCRIBER: Subscriber not Found, sending 404 \n"); t_reply("404", "Not Found"); exit; case -2: xlog("L_INFO", " TOSUBSCRIBER: Method Not allowed, sending 405 \n"); sl_send_reply("405", "Method Not Allowed"); exit; } } xlog("L_INFO", " TOSUBSCRIBER: subscriber is authorized! \n"); # not sure about any of this that follows # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { xlog("L_INFO", " TOSUBSCRIBER: setting flag FLT_ACCMISSED \n"); setflag(FLT_ACCMISSED); } xlog("L_INFO", " TOSUBSCRIBER: attempting to do: t_relay \n"); if (!t_relay()) { xlog("L_INFO", " TOSUBSCRIBER: ERROR in t_relay \n"); sl_reply_error(); }; xlog("L_INFO", " TOSUBSCRIBER: message proxied, now exiting \n"); exit; } ################################################################### # Sample failure route # One source for this block: http://pastebin.com/NMZiBuNm # A source: http://www.kamailio.org/dokuwiki/doku.php/core-cookbook:3.1.x#failure_route # 'failure_route' is executed only by TM module after it was armed via t_on_failure(“failure_route_index”) failure_route[FAIL_ONE] { xlog("L_INFO", " FAIL_ONE: starting up Sample failure route (looks for canceled)\n"); if (t_is_canceled()) { exit; } if (t_check_status("486|408")) { xlog("L_INFO","[1] ($ru) \n"); sl_send_reply("ACK", "Busy"); # send an ACK back to sender t_relay(); # forward it on } # this may do what I want ?? # if(is_method("INVITE")) { xlog("L_INFO", " leaving route[FAIL_ONE] Sample failure route \n"); } ################################################################### route[RELAY] { xlog("L_INFO", " RELAY starting up\n"); /* example how to enable some additional event routes */ if (is_method("INVITE")) { xlog("L_INFO", " RELAY: Error: INVITE sending sl_on_failure\n"); t_on_failure("FAIL_ONE"); } if (!t_relay()) { xlog("L_INFO", " RELAY: Error sending sl_reply_error\n"); sl_reply_error(); } xlog("L_INFO", " leaving [RELAY]\n"); exit; } ################################################################### # Presence server route route[PRESENCE] { xlog("L_INFO", " PRESENCE: starting up \n"); if(!is_method("PUBLISH|SUBSCRIBE")) return; # if presence enabled, this part will not be executed # if (is_method("PUBLISH") || $rU==$null) # { xlog("L_INFO", " PRESENCE: PUBLISH and SUBSCRIBE not supported sending 404\n"); sl_send_reply("404", "Not here"); exit; # } # xlog("L_INFO", "leaving route[PRESENCE] Presence server route\n"); # return; } ################################################################### # Caller NAT detection route route[NAT] { xlog("L_INFO", " NAT: NAT detection route (currently does nothing)\n"); xlog("L_INFO", " leaving route[NAT] NAT detection route \n"); return; } ################################################################### # RTPProxy control route[RTPPROXY] { xlog("L_INFO", " RTPPROXY: RTPProxy control(currently does nothing)\n"); xlog("L_INFO", " leaving route[RTPPROXY] RTPProxy control\n"); return; } ################################################################### # Routing to foreign domains route[SIPOUT] { xlog("L_INFO", " SIPOUT: Routing to foreign domains\n"); if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(RELAY); } xlog("L_INFO", " leaving route[SIPOUT] Routing to foreign domains\n"); } ################################################################### # PSTN GW routing route[PSTN] { xlog("L_INFO", " PSTN: PSTN GW routing (currently does nothing) \n"); xlog("L_INFO", " leaving route[PSTN] PSTN GW routing\n"); return; } ################################################################### # PROVIDER or our AST boxes may send an options message, they are our friends so just 200 OK them # By CEW 2Feb2011 route[HANDLEOPTIONS] { if (is_method("OPTIONS")) { # xlog("L_INFO", " HANDLEOPTIONS: found options message \n"); # if from AST if ($src_ip=~"2\.2\.2\.1") { # xlog("L_INFO", " HANDLEOPTIONS from our AST, sending 200 OK\n"); sl_send_reply("200", "ok"); exit; }; # if from PROVIDER if ($src_ip=~"2\.2\.2\.40") { xlog("L_INFO", " HANDLEOPTIONS from PROVIDER, sending 200 OK\n"); sl_send_reply("200", "ok"); exit; }; # if we made it here, just throw the message away # this assumes that none of our sip clients will send us Options messages! exit; }; }

1 0

Via branch parameter in end2end ACK
by Juha Heinanen 20 May '11

20 May '11

Daniel-Constantin Mierla writes: > if I understood right, it is about 200 ACK which are forwarded > stateless. If yes, then should be only in forward.c, the condition at > line 545, where the assignment can be changed in a memcpy. for example like this? if so, i can go and make the commit. -- juha if (syn_branch ) { memcpy*(*msg->add_to_branch_s, "z9hG4bKcydzigwkX", 16); msg->add_to_branch_len=16; } else {

2 1

IPv6
by Jon Farmer 20 May '11

20 May '11

Hi How do I represent the following if the address was IPv6? if(src_ip ==127.0.0.1) If I specify the IPv6 address with our without [] I get WARNING:core:fix_socket_list: could not rev. resolve <ipv6_addr> Regards Jon -- Jon Farmer Tel: 07795 118140 Email: viperdudeuk(a)gmail.com Twitter: @viperdudeuk

3 2

xHTTP module.
by David J. 20 May '11

20 May '11

I was wondering if I can make an HTTP request to Kamailio; and then have kamailio do a lookup based on passed parameters to connect callers. I am trying to make a click-to-dial type application; I was looking at the HTTP server inside kamailio; It was interesting to me to try to use this as an alternative to php-asterisk; (I know kamailio is not a full blown application server;) If anyone could help me with this I would appreciate it; Thanks in advance. David.

2 4

Re: [SR-Users] ACK to 486 is not being proxied
by Daniel-Constantin Mierla 20 May '11

20 May '11

Please do not write private emails, keep the mailing list cc-ed. Can you get the output of: ngrep -d any -qt -W byline port 5040 on kamailio server for such a call? Include all messages, since you may miss something in your filtering (like you did below), of course you can mask the ip addresses. Also, be sure you do t_relay() to forward the invite from kamailio on. Cheers, Daniel On 5/20/11 3:47 PM, Carl Wagner wrote: > Daniel, > > With a non-2xx reply being hop-by-hop, shouldn't Kamailio receive the > 486, send the ACK to the sender, and then proxy the 486? > > I am not seeing the 486 message in the Kamailio log so I assume that > it automatically handles it without hitting the kamailio.cfg script? > Or do I need to set up a failure_route to catch the 486 and send the > ACK there? > > Please see the log below. > > Thanks, > Carl > > > > # Asterisk to Kamailio > U 2011/05/19 14:13:01.467157 3.4.5.167:5060 -> 3.4.5.164:5060 > INVITE sip:+13031112222@3.4.5.164 SIP/2.0. > Via: SIP/2.0/UDP 3.4.5.167:5060;branch=z9hG4bK3183c9de;rport. > From: "+13032223333" <sip:+13032223333@3.4.5.167>;tag=as1f559f5a. > To: <sip:+13031112222@3.4.5.164>. > Contact: <sip:+13032223333@3.4.5.167>. > Call-ID: 37ef92a26c31a7f85958ae996d49a872(a)3.4.5.167. > CSeq: 102 INVITE. > User-Agent: Asterisk. > Max-Forwards: 70. > Date: Thu, 19 May 2011 20:13:01 GMT. > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. > Supported: replaces. > Content-Type: application/sdp. > Content-Length: 238. > . > v=0. > o=root 27806 27806 IN IP4 3.4.5.167. > s=session. > c=IN IP4 3.4.5.167. > t=0 0. > m=audio 13424 RTP/AVP 0 101. > a=rtpmap:0 PCMU/8000. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=silenceSupp:off - - - -. > a=ptime:20. > a=sendrecv. > > # Kamailio to provider > U 2011/05/19 14:13:01.468627 3.4.5.164:5060 -> 2.3.4.40:5060 > INVITE sip:+13031112222@2.3.4.40:5060 SIP/2.0. > Record-Route: <sip:3.4.5.164;lr=on>. > Via: SIP/2.0/UDP 3.4.5.164;branch=0. > Via: SIP/2.0/UDP 3.4.5.167:5060;branch=z9hG4bK3183c9de;rport=5060. > From: "+13032223333" <sip:+13032223333@3.4.5.167>;tag=as1f559f5a. > To: <sip:+13031112222@2.3.4.40:5060>. > Contact: <sip:+13032223333@3.4.5.167>. > Call-ID: 37ef92a26c31a7f85958ae996d49a872(a)3.4.5.167. > CSeq: 102 INVITE. > User-Agent: Asterisk. > Max-Forwards: 69. > Date: Thu, 19 May 2011 20:13:01 GMT. > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. > Supported: replaces. > Content-Type: application/sdp. > Content-Length: 238. > . > v=0. > o=root 27806 27806 IN IP4 3.4.5.167. > s=session. > c=IN IP4 3.4.5.167. > t=0 0. > m=audio 13424 RTP/AVP 0 101. > a=rtpmap:0 PCMU/8000. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=silenceSupp:off - - - -. > a=ptime:20. > a=sendrecv. > > > U 2011/05/19 14:13:01.511775 2.3.4.40:37897 -> 3.4.5.164:5060 > SIP/2.0 100 Trying. > Via: SIP/2.0/UDP 3.4.5.164;branch=0,SIP/2.0/UDP > 3.4.5.167:5060;branch=z9hG4bK3183c9de;rport=5060. > From: "+13032223333" <sip:+13032223333@3.4.5.167>;tag=as1f559f5a. > To: <sip:+13031112222@2.3.4.40:5060>. > Call-ID: 37ef92a26c31a7f85958ae996d49a872(a)3.4.5.167. > CSeq: 102 INVITE. > Content-Length: 0. > . > > > U 2011/05/19 14:13:01.513162 3.4.5.164:5060 -> 3.4.5.167:5060 > SIP/2.0 100 Trying. > Via: SIP/2.0/UDP 3.4.5.167:5060;branch=z9hG4bK3183c9de;rport=5060. > From: "+13032223333" <sip:+13032223333@3.4.5.167>;tag=as1f559f5a. > To: <sip:+13031112222@2.3.4.40:5060>. > Call-ID: 37ef92a26c31a7f85958ae996d49a872(a)3.4.5.167. > CSeq: 102 INVITE. > Content-Length: 0. > . > > # provider says they are busy > U 2011/05/19 14:13:03.239462 2.3.4.40:37897 -> 3.4.5.164:5060 > SIP/2.0 486 Busy Here. > Via: SIP/2.0/UDP 3.4.5.164;branch=0,SIP/2.0/UDP > 3.4.5.167:5060;branch=z9hG4bK3183c9de;rport=5060. > Record-Route: <sip:3.4.5.164;lr=on>. > From: "+13032223333" <sip:+13032223333@3.4.5.167>;tag=as1f559f5a. > To: <sip:+13031112222@2.3.4.40:5060>;tag=VPST506071629352. > Call-ID: 37ef92a26c31a7f85958ae996d49a872(a)3.4.5.167. > CSeq: 102 INVITE. > Contact: <sip:+13031112222@2.3.4.40:5060;transport=udp>. > Content-Length: 0. > . > > # > # should Kamailio send the ACK here? > # > > # Kamailio proxies the 486 to Asterisk > U 2011/05/19 14:13:03.240820 3.4.5.164:5060 -> 3.4.5.167:5060 > SIP/2.0 486 Busy Here. > Via: SIP/2.0/UDP 3.4.5.167:5060;branch=z9hG4bK3183c9de;rport=5060. > Record-Route: <sip:3.4.5.164;lr=on>. > From: "+13032223333" <sip:+13032223333@3.4.5.167>;tag=as1f559f5a. > To: <sip:+13031112222@2.3.4.40:5060>;tag=VPST506071629352. > Call-ID: 37ef92a26c31a7f85958ae996d49a872(a)3.4.5.167. > CSeq: 102 INVITE. > Contact: <sip:+13031112222@2.3.4.40:5060;transport=udp>. > Content-Length: 0. > . > > > # ACK from Asterisk - should be swallowed by Kamailio > U 2011/05/19 14:13:03.243870 3.4.5.167:5060 -> 3.4.5.164:5060 > ACK sip:+13031112222@3.4.5.164 SIP/2.0. > Via: SIP/2.0/UDP 3.4.5.167:5060;branch=z9hG4bK3183c9de;rport. > From: "+13032223333" <sip:+13032223333@3.4.5.167>;tag=as1f559f5a. > To: <sip:+13031112222@3.4.5.164>;tag=VPST506071629352. > Contact: <sip:+13032223333@3.4.5.167>. > Call-ID: 37ef92a26c31a7f85958ae996d49a872(a)3.4.5.167. > CSeq: 102 ACK. > User-Agent: Asterisk. > Max-Forwards: 70. > Content-Length: 0. > . > > > ==== a bunch of retransmitted 486's chopped ============= > > > > > > > > On 05/19/2011 02:30 AM, Daniel-Constantin Mierla wrote: >> Hello, >> >> the ACK for non-2xx replies is hop-by-hop, meaning that kamailio will >> generate one when the reply is received, will send back the reply and >> when the ack comes from upstream, it will absorb it. >> >> In this case it seems that ack is not matching any invite transaction >> and thus is not known where to send it. >> >> Can you send the invite, reply and the ack requests taken with ngrep >> for such case? Maybe is something broken in the content of the message. >> >> Cheers, >> Daniel >> >> On 5/18/11 11:23 PM, Carl Wagner wrote: >>> Hi, >>> >>> I have Kamailio set up to act as a proxy and load balancer. Most >>> things are working correctly but for some reason I can't get >>> Kamailio to proxy an ACK to a 486 Busy. >>> I assume that it is something wrong that I have done in the >>> kamailio.cfg file but I am not seeing it. >>> >>> Does the t_check_trans() return false because the call state is Busy? >>> >>> Please let me know if you need other logs or information. >>> >>> Thanks in advance, >>> Carl >>> >>> >>> >>> ================== Call flow: (from ngrep) ====================== >>> >>> Asterisk Kamailio Provider >>> INVITE---------> >>> INVITE---------> >>> <----------100 Trying >>> <---------100 Trying >>> <----------486 Busy >>> <---------486 Busy >>> ACK-------------> >>> ====== Kamailio does not proxy the ACK here ======= >>> <----------486 Busy >>> <---------486 Busy >>> ACK-------------> >>> <----------486 Busy >>> <---------486 Busy >>> >>> >>> === kamailio.cfg snippet >>> ... >>> route >>> { >>> route(REQINIT); # remove malformed messages >>> >>> # handle requests within SIP dialogs >>> route(WITHINDLG); >>> ... >>> } >>> >>> ################################################################### >>> # Handle requests within SIP dialogs (request has a TO: Tag) >>> route[WITHINDLG] >>> { >>> if (has_totag()) >>> { >>> xlog("L_INFO", " WITHINDLG: SIP Request: [$rm] ruri=[$ru] (from >>> [$fu] to [$tu], Call-ID=[$ci], CSeq=[$cs])\n"); >>> if (loose_route()) >>> { >>> ... >>> } >>> else # not loose_route >>> { >>> if ( is_method("ACK") ) >>> { >>> if ( t_check_trans() ) # see if a message is related to >>> a transaction >>> { >>> ... >>> } >>> else >>> { >>> # ACK without matching transaction ... ignore and >>> discard >>> xlog("L_INFO", " WITHINDLG: has TO: tag AND loose_route is NOT true >>> and is_method = ACK and t_check_trans=FALSE\n"); >>> >>> # not forwarded here!!! Tried both t_relay and forward. >>> # $var(a) = t_relay(); >>> $var(a) = forward(); >>> xlog ("L_INFO", " WITHINDLG: (ReturnCode = [$var(a)] >>> exiting)\n"); >>> >>> exit; >>> } >>> } >>> sl_send_reply("404","Not here"); >>> } >>> exit; >>> } >>> } >>> >>> ============= End of kamailio.cfg ================ >>> >>> >>> ============= /var/log/messages - snippit of the message ========== >>> >>> May 18 11:21:05 kam0 /usr/sbin/kamailio[21282]: INFO: <script>: >>> ======== processing new message >>> May 18 11:21:05 kam0 /usr/sbin/kamailio[21282]: INFO: <script>: >>> MAIN: SIP Request: [ACK] ruri=[sip:+13031112222@2.3.4.5] (from >>> [sip:+13031112222@3.4.5.6] to [sip:+13031112222@2.3.4.5], >>> Call-ID=[21a30b9168e0a8c963d47ca1361cef77(a)3.4.5.6]) CSeq=[102] >>> May 18 11:21:05 kam0 /usr/sbin/kamailio[21282]: INFO: <script>: >>> WITHINDLG: SIP Request: [ACK] ruri=[sip:+13031112222@2.3.4.5] (from >>> [sip:+13031112222@3.4.5.6] to [sip:+13031112222@2.3.4.5], >>> Call-ID=[21a30b9168e0a8c963d47ca1361cef77(a)3.4.5.6], CSeq=[102]) >>> May 18 11:21:05 kam0 /usr/sbin/kamailio[21282]: INFO: <script>: >>> WITHINDLG: has TO: tag >>> May 18 11:21:05 kam0 /usr/sbin/kamailio[21282]: INFO: <script>: >>> WITHINDLG: has TO: tag AND loose_route is NOT true and is_method = >>> ACK and t_check_trans=FALSE >>> May 18 11:21:05 kam0 /usr/sbin/kamailio[21282]: INFO: <script>: >>> WITHINDLG: the ACK to a 486 was not being processed so I am >>> adding t_relay here >>> May 18 11:21:05 kam0 /usr/sbin/kamailio[21282]: INFO: <script>: >>> WITHINDLG: (ReturnCode = [1] exiting) >>> >>> >>> >>> >>> >>> _______________________________________________ >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>> sr-users(a)lists.sip-router.org >>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> >> -- >> Daniel-Constantin Mierla >> http://www.asipto.com > -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

nonce reuse parameter
by Bruno Bresciani 20 May '11

20 May '11

Hi, on kamailio 3.1.2 the one_time_nonce parameter of the auth module is similar to the nonce_reuse parameter present on kamailio 1.5.0? I want to disable the check of the nonce reuse in kamailio 3.1.2... regards

2 2

Reactivating a failed dispatcher destination
by Lee Archer 20 May '11

20 May '11

Hi all I wonder if someone can help by pointing me in the right direction. I have dispatcher set to ping and set any failed gateways to probing but how do you get them out of this state when they start working again? I have run some tests and I must be missing something as active servers seem to stay in probing mode. I am using Kamailio 3.0.4. Thanks Lee

3 10

packaging 3.1.4
by Daniel-Constantin Mierla 20 May '11

20 May '11

Hello, I am thinking of packaging v3.1.4 next week on Thursday -- last minor release was more than 1.5month ago. Any comments against it? Cheers, Daniel -- Daniel-Constantin Mierla -- http://www.asipto.com http://linkedin.com/in/miconda -- http://twitter.com/miconda

1 0

Forward on unanswered call
by Andrew O. Zhukov 20 May '11

20 May '11

Right now I'm try to add the subj feature. Linksys/SPA2102-3.3.6 (just from the box) respond SIP/2.0 480 Temporarily not available Landline over Asterisk with E1: SIP/2.0 603 Limit exceeded Ether I so unlucky, or just forget it and do not provide it to the customers? If so, notes to developers let's take off 408 code processing from sample config. Thanks in advance for your opinion. Andrew.

2 1

Via branch parameter in end2end ACK
by Francesco Castellano 19 May '11

19 May '11

Dear sirs, using version 3.1.3 (kamailio flavour), I noticed that the Via inserted by the proxy, relaying an end2end ACK, contains a branch parameter equal to 0 (with no magic cookie prefixed). No misbehaviour followed, but I was wonder about the reason for this behaviour and whether it can be a fault of my config: route { [some checks] if (!is_method("REGISTER|MESSAGE|PUBLISH")) { record_route(); } else if (is_method("REGISTER")) { route(HANDLE_REGISTER); exit; } if (loose_route()){ if (!has_totag()){ if (!is_method("ACK")){ send_reply("403", "Preset Route Set Not Allowed"); } exit; } t_check_trans(); if (is_method("ACK")){ if ($du == $null){ handle_ruri_alias(); } if (isflagset(T_UAC_NATED) && isflagset(T_UAC_USER)){ add_contact_alias(); } remove_hf("Proxy-Authorization"); t_relay(); exit; } } [other processing] } Here's an example for the forwarded ACK: U 2011/05/17 15:05:49.392201 proxy_1_ip:5060 -> location_callee_ip:5060 ACK sip:5303838@192.168.130.169;user=phone SIP/2.0. Record-Route: <sip:proxy_1_ip;lr=on;ftag=a2z0myc9r6>. Via: SIP/2.0/UDP proxy_1_ip;branch=0. Via: SIP/2.0/UDP proxy_0_ip;rport=5060;branch=0. Via: SIP/2.0/UDP 192.168.130.171:2048;received=location_caller_ip;branch=z9hG4bK-pud4cfiz8vsh;rport=4791. From: "cento tondo" <sip:100@proxy_0_hostname>;tag=a2z0myc9r6. To: <sip:5375485@proxy_1_hostname;user=phone>;tag=00221b79b1b28969. Call-ID: 3c352c0b8419-9tbq023tzfkp. CSeq: 2 ACK. Max-Forwards: 68. Contact: <sip:100@192.168.130.171:2048;alias=location_caller_ip~4791~1;line=gwwqwi25>;reg-id=1. Content-Length: 0. . where proxy_1 does force_rport and record_route() on ACK too, and proxy_0 does not. Both proxies run the same version, and a similar ACK script processing. Best regards, Francesco Castellano

3 5

ACK to 486 is not being proxied
by Carl Wagner 19 May '11

19 May '11

Hi, I have Kamailio set up to act as a proxy and load balancer. Most things are working correctly but for some reason I can't get Kamailio to proxy an ACK to a 486 Busy. I assume that it is something wrong that I have done in the kamailio.cfg file but I am not seeing it. Does the t_check_trans() return false because the call state is Busy? Please let me know if you need other logs or information. Thanks in advance, Carl ================== Call flow: (from ngrep) ====================== Asterisk Kamailio Provider INVITE---------> INVITE---------> <----------100 Trying <---------100 Trying <----------486 Busy <---------486 Busy ACK-------------> ====== Kamailio does not proxy the ACK here ======= <----------486 Busy <---------486 Busy ACK-------------> <----------486 Busy <---------486 Busy === kamailio.cfg snippet ... route { route(REQINIT); # remove malformed messages # handle requests within SIP dialogs route(WITHINDLG); ... } ################################################################### # Handle requests within SIP dialogs (request has a TO: Tag) route[WITHINDLG] { if (has_totag()) { xlog("L_INFO", " WITHINDLG: SIP Request: [$rm] ruri=[$ru] (from [$fu] to [$tu], Call-ID=[$ci], CSeq=[$cs])\n"); if (loose_route()) { ... } else # not loose_route { if ( is_method("ACK") ) { if ( t_check_trans() ) # see if a message is related to a transaction { ... } else { # ACK without matching transaction ... ignore and discard xlog("L_INFO", " WITHINDLG: has TO: tag AND loose_route is NOT true and is_method = ACK and t_check_trans=FALSE\n"); # not forwarded here!!! Tried both t_relay and forward. # $var(a) = t_relay(); $var(a) = forward(); xlog ("L_INFO", " WITHINDLG: (ReturnCode = [$var(a)] exiting)\n"); exit; } } sl_send_reply("404","Not here"); } exit; } } ============= End of kamailio.cfg ================ ============= /var/log/messages - snippit of the message ========== May 18 11:21:05 kam0 /usr/sbin/kamailio[21282]: INFO: <script>: ======== processing new message May 18 11:21:05 kam0 /usr/sbin/kamailio[21282]: INFO: <script>: MAIN: SIP Request: [ACK] ruri=[sip:+13031112222@2.3.4.5] (from [sip:+13031112222@3.4.5.6] to [sip:+13031112222@2.3.4.5], Call-ID=[21a30b9168e0a8c963d47ca1361cef77(a)3.4.5.6]) CSeq=[102] May 18 11:21:05 kam0 /usr/sbin/kamailio[21282]: INFO: <script>: WITHINDLG: SIP Request: [ACK] ruri=[sip:+13031112222@2.3.4.5] (from [sip:+13031112222@3.4.5.6] to [sip:+13031112222@2.3.4.5], Call-ID=[21a30b9168e0a8c963d47ca1361cef77(a)3.4.5.6], CSeq=[102]) May 18 11:21:05 kam0 /usr/sbin/kamailio[21282]: INFO: <script>: WITHINDLG: has TO: tag May 18 11:21:05 kam0 /usr/sbin/kamailio[21282]: INFO: <script>: WITHINDLG: has TO: tag AND loose_route is NOT true and is_method = ACK and t_check_trans=FALSE May 18 11:21:05 kam0 /usr/sbin/kamailio[21282]: INFO: <script>: WITHINDLG: the ACK to a 486 was not being processed so I am adding t_relay here May 18 11:21:05 kam0 /usr/sbin/kamailio[21282]: INFO: <script>: WITHINDLG: (ReturnCode = [1] exiting)

2 1

redirect implementation with dispatcher
by alex pappas 19 May '11

19 May '11

Dear all, I'm trying to create a service which when a call reach Kamailio, this call get's a number of gateways in round robin mode and from them I create uri and append_branches. The all scenario works fine except that I don't get from Dispatcher the round robin mode. Is is the way I ask Dispatcher and i don't get the results I need. In other scenarios Dispatcher works fine. The other thing is that I tried the random alg of Dispatcher and that works. I'm trying to understand what I'm doing wrong and why random alg works but round robin not. I'm attaching my code if anyone has an idea of what I'm doing wrong. Thank you Alex *Scenario description:* * Call enter --> Dialplan (Here if has a match get the attributes which is a list of dispatcher gw group in the form of 100.101.#) --> Dispatcher (For every dispatcher gw group appnd the branch) --> Redirect the call* route[TEST] { if(dp_translate("$avp(s:lcr_id)", "$rU/$rU")) { $var(i) = 0; while($(var(dpattrs){s.select,$var(i),.})!="#") *#attr example is gw group of disptcher in order 100.101.# or in my scenario is 200.# which in Dispatcher is 2 gw* *10.0.0.2 and 10.0.0.3* { $avp(s:disp_dstgrp) = $(var(dpattrs){s.select,$var(i),.}{ s.int}); if(ds_select_domain("$avp(s:disp_dstgrp)", "4")) { append_branch(); revert_uri(); $avp(s:disp_dstgrp) = "null"; } $var(i) = $var(i) + 1; # for the while loop } ####################### REDIRECTION ##################### sl_send_reply("300", "Multiple Choice"); ######################################################### exit; } else { xlog(" ----- No routing found for this call -----"); } }

2 3

Kamailio 1.5 acc on glare BYE
by Tobias Lindgren 18 May '11

18 May '11

Hi all! I have a short question with regards to how the acc module handles (or rather logs) BYEs. As I understand, a BYE followed by an 200 OK will be logged by Kamailio's acc module. This works perfectly fine in my set up. However, one of the carriers that I'm using sometimes respond with a "487 Request Terminated" to BYE messages, stating that "Glare Bye condition encountered". I'm not really sure why this occurs at my carriers', but whenever this happens, acc does not log the BYE message. To my question; what happens in the acc module if sip_code > 200 is received? And, should these BYE's be logged, even though the code is 487? I'm on Kamailio 1.5 with acc logging to MySQL. Br, /Tobias

2 2

403 Proxy forbidden replay for invite that send sip gateway
by hala alramli 18 May '11

18 May '11

Hello every body,could you help me ? i bought Geographical DID numbers from gateway provider , to receive calls from pstn my problem that when gateway send invite my proxy send 403 forbidden ,i dont know why what is wrong ,any help please. this is a track for th gateway invite. my proxy ip:79.12.254.153gateway IP :77.239.128.10my geographical number :0698962030 INVITE sip:0698962030@79.12.254.153:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 77.239.128.10:5060;branch=z9hG4bKf89p48m8dkuco1tumbnfaemc44 From: <sip:0289089500@10.39.1.16:5060;user=phone>;tag=SDml08d01--45026-1f4d3a5-203bbf09-1f4d3a5 To: <sip:0698962030@10.39.6.207:5060;user=phone> Call-ID: SDml08d01-daae53b1c06b7367e53d68ad67a63d13-ao92061 CSeq: 1 INVITE User-Agent: CS2000_NGSS/8.0 Remote-Party-ID: <sip:0289089500@10.39.1.16;user=phone>; party=calling; privacy=off; screen=yes Allow: ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK P-Asserted-Identity: <sip:0289089500@10.39.1.16;user=phone> Max-Forwards: 69 Contact: <sip:77.239.128.10:5060;transport=UDP> Supported: 100rel Content-Type: application/sdp Content-Length: 236 v=0 o=PVG 1305215180830 1305215180830 IN IP4 77.239.128.10 s=- p=+1 6135555555 c=IN IP4 77.239.128.10 t=0 0 m=audio 49264 RTP/AVP 18 8 0 13 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=fmtp:18 annexb=no SIP/2.0 403 Proxy forbidden Via: SIP/2.0/UDP 77.239.128.10:5060;branch=z9hG4bKf89p48m8dkuco1tumbnfaemc44;rport=5060;received=77.239.128.10 From: <sip:0289089500@10.39.1.16:5060;user=phone>;tag=SDml08d01--45026-1f4d3a5-203bbf09-1f4d3a5 To: <sip:0698962030@10.39.6.207:5060;user=phone>;tag=4471d251144740103ef2d5e5631bdc66.799c Call-ID: SDml08d01-daae53b1c06b7367e53d68ad67a63d13-ao92061 CSeq: 1 INVITE Server: Kamailio (1.4.4-notls (x86_64/linux)) Content-Length: 0 ACK sip:0698962030@79.12.254.153:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 77.239.128.10:5060;branch=z9hG4bKf89p48m8dkuco1tumbnfaemc44;rport=5060;received=77.239.128.10 From: <sip:0289089500@10.39.1.16:5060;user=phone>;tag=SDml08d01--45026-1f4d3a5-203bbf09-1f4d3a5 To: <sip:0698962030@10.39.6.207:5060;user=phone>;tag=4471d251144740103ef2d5e5631bdc66.799c Call-ID: SDml08d01-daae53b1c06b7367e53d68ad67a63d13-ao92061 CSeq: 1 ACK Max-Forwards: 69 Content-Length: 0 INVITE sip:0698962030@79.12.254.153:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 77.239.128.10:5060;branch=z9hG4bK4t5g11q75ccl8k5vnbgabnlct2 From: <sip:026683011@10.39.1.16:5060;user=phone>;tag=SDa7sn401--45026-1f4d3ad-69943699-1f4d3ad To: <sip:0698962030@10.39.6.207:5060;user=phone> Call-ID: SDa7sn401-c505e58e02da5e4449475a8d28dbcb07-ao92061 CSeq: 1 INVITE User-Agent: CS2000_NGSS/8.0 Remote-Party-ID: <sip:026683011@10.39.1.16;user=phone>; party=calling; privacy=off; screen=yes Allow: ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK P-Asserted-Identity: <sip:026683011@10.39.1.16;user=phone> Max-Forwards: 69 Contact: <sip:77.239.128.10:5060;transport=UDP> Supported: 100rel Content-Type: application/sdp Content-Length: 236 v=0 o=PVG 1305215458700 1305215458700 IN IP4 77.239.128.10 s=- p=+1 6135555555 c=IN IP4 77.239.128.10 t=0 0 m=audio 41892 RTP/AVP 18 8 0 13 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=fmtp:18 annexb=no SIP/2.0 403 Proxy forbidden Via: SIP/2.0/UDP 77.239.128.10:5060;branch=z9hG4bK4t5g11q75ccl8k5vnbgabnlct2;rport=5060;received=77.239.128.10 From: <sip:026683011@10.39.1.16:5060;user=phone>;tag=SDa7sn401--45026-1f4d3ad-69943699-1f4d3ad To: <sip:0698962030@10.39.6.207:5060;user=phone>;tag=4471d251144740103ef2d5e5631bdc66.fe6e Call-ID: SDa7sn401-c505e58e02da5e4449475a8d28dbcb07-ao92061 CSeq: 1 INVITE Server: Kamailio (1.4.4-notls (x86_64/linux)) Content-Length: 0 sip:0698962030@79.12.254.153:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 77.239.128.10:5060;branch=z9hG4bK4t5g11q75ccl8k5vnbgabnlct2;rport=5060;received=77.239.128.10 From: <sip:026683011@10.39.1.16:5060;user=phone>;tag=SDa7sn401--45026-1f4d3ad-69943699-1f4d3ad To: <sip:0698962030@10.39.6.207:5060;user=phone>;tag=4471d251144740103ef2d5e5631bdc66.fe6e Call-ID: SDa7sn401-c505e58e02da5e4449475a8d28dbcb07-ao92061 CSeq: 1 ACK Max-Forwards: 69 Content-Length: 0

2 2

setbflag() documentation missing?
by Juha Heinanen 17 May '11

17 May '11

setbflag() is not documented in core cookbook. is it missing or somewhere else? -- juha

2 2

report from linuxtag, presentations
by Henning Westerholt 17 May '11

17 May '11

Hello, just to give you a quick feedback about the this year linuxtag - it was overall a really nice event in my opinion. A lot of people passed by at our booth to discuss about Kamailio and VoIP in general. Carsten Bock presented about our new IMS extensions, Andreas Granig showed a demo of their new kamailio based carrier product. I hope that we can provide some photos of the event soon. We've did two presentations, one about "Linux at 1&1" including a Kamailio part on wednesday, and Carsten did two "FastForward" talks on wednesday and thursday. I've uploaded the slides to our webserver: http://www.kamailio.org/events/2011-linuxtag/linux-at-1and1.pdf http://www.kamailio.org/events/2011-linuxtag/kamailio-quick.pdf Also interesting, the German Federal Office for Information Security (BSI) presented the upcoming "Gemeinschaft" PBX project version 4 that will use Kamailio (and Freeswitch). They sponsor the project, it aims to provide a secure, scalable and easy to use open source PBX plattform. Best regards, Henning

1 0

Kamailio 3.1.3 uncontrolled core dump
by Dominguez Jover, Ricardo 17 May '11

17 May '11

Hi everybody. As posted in http://lists.sip-router.org/pipermail/sr-users/2011-May/068514.html we are having an uncontrolled core dump. It seems that the crash occurs when some users registers to Kamailio. After upgrading to Kamilio release 3.1.3 we are able to generate the following GDB: GNU gdb (GDB) Red Hat Enterprise Linux (7.0.1-32.el5_6.2) Copyright (C) 2009 Free Software Foundation, Inc. License GPLv3+: GNU GPL version 3 or later <http://gnu.org/licenses/gpl.html> This is free software: you are free to change and redistribute it. There is NO WARRANTY, to the extent permitted by law. Type "show copying" and "show warranty" for details. This GDB was configured as "i386-redhat-linux-gnu". For bug reporting instructions, please see: <http://www.gnu.org/software/gdb/bugs/>... Reading symbols from /usr/local/kamailio-3.1.3/sbin/kamailio...done. Reading symbols from /lib/libdl.so.2...(no debugging symbols found)...done. Loaded symbols for /lib/libdl.so.2 Reading symbols from /lib/libresolv.so.2...(no debugging symbols found)...done. Loaded symbols for /lib/libresolv.so.2 Reading symbols from /lib/libc.so.6...(no debugging symbols found)...done. Loaded symbols for /lib/libc.so.6 Reading symbols from /lib/ld-linux.so.2...(no debugging symbols found)...done. Loaded symbols for /lib/ld-linux.so.2 Reading symbols from /usr/local/kamailio-3.1.3/lib/kamailio/modules/db_mysql.so...done. Loaded symbols for /usr/local/kamailio-3.1.3/lib/kamailio/modules/db_mysql.so Reading symbols from /usr/lib/mysql/libmysqlclient.so.15...(no debugging symbols found)...done. Loaded symbols for /usr/lib/mysql/libmysqlclient.so.15 Reading symbols from /usr/lib/libz.so.1...(no debugging symbols found)...done. Loaded symbols for /usr/lib/libz.so.1 Reading symbols from /lib/libcrypt.so.1...(no debugging symbols found)...done. Loaded symbols for /lib/libcrypt.so.1 Reading symbols from /lib/libnsl.so.1...(no debugging symbols found)...done. Loaded symbols for /lib/libnsl.so.1 Reading symbols from /lib/libm.so.6...(no debugging symbols found)...done. Loaded symbols for /lib/libm.so.6 Reading symbols from /lib/libssl.so.6...(no debugging symbols found)...done. Loaded symbols for /lib/libssl.so.6 Reading symbols from /lib/libcrypto.so.6...(no debugging symbols found)...done. Loaded symbols for /lib/libcrypto.so.6 Reading symbols from /usr/local/kamailio-3.1.3/lib/kamailio/libsrdb2.so.1...done. Loaded symbols for /usr/local/kamailio-3.1.3/lib/kamailio/libsrdb2.so.1 Reading symbols from /usr/local/kamailio-3.1.3/lib/kamailio/libsrdb1.so.1...done. Loaded symbols for /usr/local/kamailio-3.1.3/lib/kamailio/libsrdb1.so.1 Reading symbols from /usr/lib/libgssapi_krb5.so.2...(no debugging symbols found)...done. Loaded symbols for /usr/lib/libgssapi_krb5.so.2 Reading symbols from /usr/lib/libkrb5.so.3...(no debugging symbols found)...done. Loaded symbols for /usr/lib/libkrb5.so.3 Reading symbols from /lib/libcom_err.so.2...(no debugging symbols found)...done. Loaded symbols for /lib/libcom_err.so.2 Reading symbols from /usr/lib/libk5crypto.so.3...(no debugging symbols found)...done. Loaded symbols for /usr/lib/libk5crypto.so.3 Reading symbols from /usr/lib/libkrb5support.so.0...(no debugging symbols found)...done. Loaded symbols for /usr/lib/libkrb5support.so.0 Reading symbols from /lib/libkeyutils.so.1...(no debugging symbols found)...done. Loaded symbols for /lib/libkeyutils.so.1 Reading symbols from /lib/libselinux.so.1...(no debugging symbols found)...done. Loaded symbols for /lib/libselinux.so.1 Reading symbols from /lib/libsepol.so.1...(no debugging symbols found)...done. Loaded symbols for /lib/libsepol.so.1 Reading symbols from /usr/local/kamailio-3.1.3/lib/kamailio/modules_k/mi_fifo.so...done. Loaded symbols for /usr/local/kamailio-3.1.3/lib/kamailio/modules_k/mi_fifo.so Reading symbols from /usr/local/kamailio-3.1.3/lib/kamailio/libkmi.so.1...done. Loaded symbols for /usr/local/kamailio-3.1.3/lib/kamailio/libkmi.so.1 Reading symbols from /usr/local/kamailio-3.1.3/lib/kamailio/modules_k/kex.so...done. Loaded symbols for /usr/local/kamailio-3.1.3/lib/kamailio/modules_k/kex.so Reading symbols from /usr/local/kamailio-3.1.3/lib/kamailio/libkcore.so.1...done. Loaded symbols for /usr/local/kamailio-3.1.3/lib/kamailio/libkcore.so.1 Reading symbols from /usr/local/kamailio-3.1.3/lib/kamailio/modules/tm.so...done. Loaded symbols for /usr/local/kamailio-3.1.3/lib/kamailio/modules/tm.so Reading symbols from /usr/local/kamailio-3.1.3/lib/kamailio/modules_k/tmx.so...done. Loaded symbols for /usr/local/kamailio-3.1.3/lib/kamailio/modules_k/tmx.so Reading symbols from /usr/local/kamailio-3.1.3/lib/kamailio/modules/sl.so...done. Loaded symbols for /usr/local/kamailio-3.1.3/lib/kamailio/modules/sl.so Reading symbols from /usr/local/kamailio-3.1.3/lib/kamailio/modules_k/rr.so...done. Loaded symbols for /usr/local/kamailio-3.1.3/lib/kamailio/modules_k/rr.so Reading symbols from /usr/local/kamailio-3.1.3/lib/kamailio/modules_k/pv.so...done. Loaded symbols for /usr/local/kamailio-3.1.3/lib/kamailio/modules_k/pv.so Reading symbols from /usr/local/kamailio-3.1.3/lib/kamailio/modules_k/maxfwd.so...done. Loaded symbols for /usr/local/kamailio-3.1.3/lib/kamailio/modules_k/maxfwd.so Reading symbols from /usr/local/kamailio-3.1.3/lib/kamailio/modules_k/usrloc.so...done. Loaded symbols for /usr/local/kamailio-3.1.3/lib/kamailio/modules_k/usrloc.so Reading symbols from /usr/local/kamailio-3.1.3/lib/kamailio/modules_k/registrar.so...done. Loaded symbols for /usr/local/kamailio-3.1.3/lib/kamailio/modules_k/registrar.so Reading symbols from /usr/local/kamailio-3.1.3/lib/kamailio/modules_k/textops.so...done. Loaded symbols for /usr/local/kamailio-3.1.3/lib/kamailio/modules_k/textops.so Reading symbols from /usr/local/kamailio-3.1.3/lib/kamailio/modules_k/siputils.so...done. Loaded symbols for /usr/local/kamailio-3.1.3/lib/kamailio/modules_k/siputils.so Reading symbols from /usr/local/kamailio-3.1.3/lib/kamailio/modules_k/xlog.so...done. Loaded symbols for /usr/local/kamailio-3.1.3/lib/kamailio/modules_k/xlog.so Reading symbols from /usr/local/kamailio-3.1.3/lib/kamailio/modules/sanity.so...done. Loaded symbols for /usr/local/kamailio-3.1.3/lib/kamailio/modules/sanity.so Reading symbols from /usr/local/kamailio-3.1.3/lib/kamailio/modules/ctl.so...done. Loaded symbols for /usr/local/kamailio-3.1.3/lib/kamailio/modules/ctl.so Reading symbols from /usr/local/kamailio-3.1.3/lib/kamailio/modules/mi_rpc.so...done. Loaded symbols for /usr/local/kamailio-3.1.3/lib/kamailio/modules/mi_rpc.so Reading symbols from /usr/local/kamailio-3.1.3/lib/kamailio/modules_k/acc.so...done. Loaded symbols for /usr/local/kamailio-3.1.3/lib/kamailio/modules_k/acc.so Reading symbols from /usr/local/kamailio-3.1.3/lib/kamailio/modules/auth.so...done. Loaded symbols for /usr/local/kamailio-3.1.3/lib/kamailio/modules/auth.so Reading symbols from /usr/local/kamailio-3.1.3/lib/kamailio/modules_k/auth_db.so...done. Loaded symbols for /usr/local/kamailio-3.1.3/lib/kamailio/modules_k/auth_db.so Reading symbols from /usr/local/kamailio-3.1.3/lib/kamailio/modules_k/permissions.so...done. Loaded symbols for /usr/local/kamailio-3.1.3/lib/kamailio/modules_k/permissions.so Reading symbols from /usr/local/kamailio-3.1.3/lib/kamailio/modules_k/alias_db.so...done. Loaded symbols for /usr/local/kamailio-3.1.3/lib/kamailio/modules_k/alias_db.so Reading symbols from /usr/local/kamailio-3.1.3/lib/kamailio/modules_k/domain.so...done. Loaded symbols for /usr/local/kamailio-3.1.3/lib/kamailio/modules_k/domain.so Reading symbols from /usr/local/kamailio-3.1.3/lib/kamailio/modules_k/nathelper.so...done. Loaded symbols for /usr/local/kamailio-3.1.3/lib/kamailio/modules_k/nathelper.so Reading symbols from /usr/local/kamailio-3.1.3/lib/kamailio/modules_k/rtpproxy.so...done. Loaded symbols for /usr/local/kamailio-3.1.3/lib/kamailio/modules_k/rtpproxy.so Reading symbols from /usr/local/kamailio-3.1.3/lib/kamailio/modules/tls.so...done. Loaded symbols for /usr/local/kamailio-3.1.3/lib/kamailio/modules/tls.so Reading symbols from /lib/libnss_files.so.2...(no debugging symbols found)...done. Loaded symbols for /lib/libnss_files.so.2 Reading symbols from /usr/local/kamailio-3.1.3/lib/kamailio/modules_k/sqlops.so...done. Loaded symbols for /usr/local/kamailio-3.1.3/lib/kamailio/modules_k/sqlops.so Reading symbols from /usr/local/kamailio-3.1.3/lib/kamailio/modules_k/group.so...done. Loaded symbols for /usr/local/kamailio-3.1.3/lib/kamailio/modules_k/group.so Reading symbols from /usr/local/kamailio-3.1.3/lib/kamailio/modules/dialplan.so...done. Loaded symbols for /usr/local/kamailio-3.1.3/lib/kamailio/modules/dialplan.so Reading symbols from /lib/libpcre.so.0...(no debugging symbols found)...done. Loaded symbols for /lib/libpcre.so.0 Reading symbols from /lib/libnss_dns.so.2...(no debugging symbols found)...done. Loaded symbols for /lib/libnss_dns.so.2 Core was generated by `/usr/local/kamailio-3.1.3/sbin/kamailio -P /var/run/kamailio.pid -m 64 -u kamai'. Program terminated with signal 11, Segmentation fault. #0 pv_set_ruri_user (msg=0x83622c8, param=0x8320078, op=254, val=0xbfbaa83c) at pv_core.c:1756 1756 val->rs.s[val->rs.len] = '\0'; (gdb) (gdb) (gdb) exit (gdb) quit Last debug messages before the crash are: May 13 09:32:06 tip1 /usr/local/kamailio-3.1.3/sbin/kamailio[3044]: ERROR: <core> [resolve.c:1540]: ERROR: sip_hostport2su: could not resolve hostname: "@192.168.64.36" May 13 09:32:06 tip1 /usr/local/kamailio-3.1.3/sbin/kamailio[3044]: ERROR: tm [ut.h:318]: failed to resolve "@192.168.64.36" May 13 09:32:06 tip1 /usr/local/kamailio-3.1.3/sbin/kamailio[3044]: ERROR: tm [t_fwd.c:1528]: ERROR: t_forward_nonack: failure to add branches May 13 09:32:06 tip1 /usr/local/kamailio-3.1.3/sbin/kamailio[3044]: ERROR: sl [sl_funcs.c:282]: ERROR: sl_reply_error used: Unresolvable destination (478/SL) May 13 09:32:14 tip1 /usr/local/kamailio-3.1.3/sbin/kamailio[3043]: ERROR: <core> [resolve.c:1540]: ERROR: sip_hostport2su: could not resolve hostname: "@192.168.64.36" May 13 09:32:14 tip1 /usr/local/kamailio-3.1.3/sbin/kamailio[3043]: ERROR: tm [ut.h:318]: failed to resolve "@192.168.64.36" May 13 09:32:14 tip1 /usr/local/kamailio-3.1.3/sbin/kamailio[3043]: ERROR: tm [t_fwd.c:1528]: ERROR: t_forward_nonack: failure to add branches May 13 09:32:14 tip1 /usr/local/kamailio-3.1.3/sbin/kamailio[3043]: ERROR: sl [sl_funcs.c:282]: ERROR: sl_reply_error used: Unresolvable destination (478/SL) May 13 09:32:16 tip1 /usr/local/kamailio-3.1.3/sbin/kamailio[3047]: : <core> [pass_fd.c:293]: ERROR: receive_fd: EOF on 21 May 13 09:32:16 tip1 /usr/local/kamailio-3.1.3/sbin/kamailio[3047]: ERROR: <core> [tcp_main.c:4315]: connect 192.168.67.21:49320 failed (timeout) May 13 09:32:17 tip1 /usr/local/kamailio-3.1.3/sbin/kamailio[3047]: : <core> [pass_fd.c:293]: ERROR: receive_fd: EOF on 21 May 13 09:32:17 tip1 /usr/local/kamailio-3.1.3/sbin/kamailio[3047]: ERROR: <core> [io_wait.h:628]: ERROR: io_watch_del: trying to delete already erased entry 21 in the hash(-1, 0, 0xb40a02e4) flags 0) May 13 09:32:17 tip1 /usr/local/kamailio-3.1.3/sbin/kamailio[3047]: ERROR: <core> [io_wait.h:628]: ERROR: io_watch_del: trying to delete already erased entry 23 in the hash(-1, 0, 0x83657ec) flags 0) May 13 09:32:17 tip1 /usr/local/kamailio-3.1.3/sbin/kamailio[3047]: : <core> [pass_fd.c:209]: ERROR: send_fd: sendmsg failed sending 36 on 23: Broken pipe (32) May 13 09:32:17 tip1 /usr/local/kamailio-3.1.3/sbin/kamailio[3047]: ERROR: <core> [tcp_main.c:3917]: ERROR: send2child: send_fd failed for 0xb421c7e4 (flags 0x4018), fd 36 May 13 09:32:16 tip1 /usr/local/kamailio-3.1.3/sbin/kamailio[3027]: ALERT: <core> [main.c:741]: child process 3043 exited by a signal 11 May 13 09:32:17 tip1 /usr/local/kamailio-3.1.3/sbin/kamailio[3027]: ALERT: <core> [main.c:744]: core was generated May 13 09:32:17 tip1 /usr/local/kamailio-3.1.3/sbin/kamailio[3027]: INFO: <core> [main.c:756]: INFO: terminating due to SIGCHLD May 13 09:32:17 tip1 /usr/local/kamailio-3.1.3/sbin/kamailio[3047]: INFO: <core> [main.c:807]: INFO: signal 15 received May 13 09:32:17 tip1 /usr/local/kamailio-3.1.3/sbin/kamailio[3046]: INFO: <core> [main.c:807]: INFO: signal 15 received May 13 09:32:17 tip1 /usr/local/kamailio-3.1.3/sbin/kamailio[3045]: INFO: <core> [main.c:807]: INFO: signal 15 received May 13 09:32:17 tip1 /usr/local/kamailio-3.1.3/sbin/kamailio[3044]: INFO: <core> [main.c:807]: INFO: signal 15 received May 13 09:32:17 tip1 /usr/local/kamailio-3.1.3/sbin/kamailio[3039]: INFO: <core> [main.c:807]: INFO: signal 15 received May 13 09:32:17 tip1 /usr/local/kamailio-3.1.3/sbin/kamailio[3035]: INFO: <core> [main.c:807]: INFO: signal 15 received May 13 09:32:17 tip1 /usr/local/kamailio-3.1.3/sbin/kamailio[3034]: INFO: <core> [main.c:807]: INFO: signal 15 received May 13 09:32:17 tip1 /usr/local/kamailio-3.1.3/sbin/kamailio[3042]: INFO: <core> [main.c:807]: INFO: signal 15 received May 13 09:32:17 tip1 /usr/local/kamailio-3.1.3/sbin/kamailio[3041]: INFO: <core> [main.c:807]: INFO: signal 15 received May 13 09:32:17 tip1 /usr/local/kamailio-3.1.3/sbin/kamailio[3040]: INFO: <core> [main.c:807]: INFO: signal 15 received May 13 09:32:17 tip1 /usr/local/kamailio-3.1.3/sbin/kamailio[3038]: INFO: <core> [main.c:807]: INFO: signal 15 received May 13 09:32:17 tip1 /usr/local/kamailio-3.1.3/sbin/kamailio[3037]: INFO: <core> [main.c:807]: INFO: signal 15 received May 13 09:32:24 tip1 hb_standby[3440]: Going standby [foreign]. May 13 09:32:25 tip1 /usr/local/kamailio-3.1.3/sbin/kamailio[3027]: ERROR: ctl [ctl.c:379]: ERROR: ctl: could not delete unix socket /tmp/ser_ctl: Operation not permitted (1) May 13 09:32:25 tip1 heartbeat: [2048]: info: tip1.umh.es wants to go standby [foreign] May 13 09:32:36 tip1 heartbeat: [2048]: WARN: No reply to standby request. Standby request cancelled. Any help would be appreciated. Cheers, Ricardo Dominguez

2 3

Does t_newtran() behave different than in Kamailio 1.5?
by Iñaki Baz Castillo 17 May '11

17 May '11

Hi, as Kamailio 1.5 TM documentation states, changes to transaction (adding headers, filling AVP's and so) after calling t_newtran() are not saved into the transaction: http://kamailio.org/docs/modules/1.5.x/tm.html#id2510920 NOTE that the changes on the request that are made after this function call will not be saved into transaction!!! However in 3.X TM documentation such note is not present: http://kamailio.org/docs/modules/devel/modules/tm.html#t_newtran So, does 3.X behave different and changes after t_newtran are added to the transaction or not? Thanks a lot. -- Iñaki Baz Castillo <ibc(a)aliax.net>

4 8

core in dialog module
by Anton Roman 17 May '11

17 May '11

Hi all, we got a core in dialog module. We are using kamailio 3.1.2. Below you can find a full backtrace from the dump and the Kamailio compilation options. Please, if you need further information don't hesitate to ask me for it. I can't precise the situation when it is generated because we have a quite high load in this server. Thanks in advance. Antón (gdb) bt full #0 unref_dlg (dlg=0x7f08a9f67da8, cnt=1) at dlg_hash.c:598 d_entry = (struct dlg_entry *) 0x7f10304b8b68 #1 0x00007f08ce92fa02 in run_trans_callbacks_internal (cb_lst=0x7f08aa203e98, type=32768, trans=0x7f08aa203e28, params=0x7fff49059a10) at t_hooks.c:290 cbp = (struct tm_callback *) 0x7f08a9f6e7e0 backup_from = (avp_list_t *) 0x8b3330 backup_to = (avp_list_t *) 0x8b3338 backup_dom_from = (avp_list_t *) 0x8b3340 backup_dom_to = (avp_list_t *) 0x8b3348 backup_uri_from = (avp_list_t *) 0x8b3320 backup_uri_to = (avp_list_t *) 0x8b3328 #2 0x00007f08ce92fc56 in run_trans_callbacks (type=32768, trans=<value optimized out>, req=0x1, rpl=0x7f10304b8b68, code=-868566200) at t_hooks.c:317 params = {req = 0x0, rpl = 0x0, param = 0x7f08a9f6e7f0, code = 0, flags = 0, branch = 0, t_rbuf = 0x0, dst = 0x0, send_buf = { s = 0x0, len = 0}} #3 0x00007f08ce915b36 in free_cell (dead_cell=0x7f08aa203e28) at h_table.c:136 b = <value optimized out> i = <value optimized out> rpl = <value optimized out> tt = <value optimized out> foo = <value optimized out> cbs = <value optimized out> ---Type <return> to continue, or q <return> to quit--- __FUNCTION__ = "free_cell" #4 0x00007f08ce9319f1 in wait_handler (ti=<value optimized out>, wait_tl=<value optimized out>, data=<value optimized out>) at timer.c:645 p_cell = (struct cell *) 0x7f08aa203e28 #5 0x0000000000513d8f in timer_main () at timer.c:894 No locals. #6 0x000000000046501b in main_loop () at main.c:1618 i = 4 pid = <value optimized out> si = (struct socket_info *) 0x0 si_desc = "udp receiver child=3 sock=XXX.XXX.XXX.XX:XXXX\000\000\000\210�\231\000\000\000\000\000\031", '\0' <repeats 15 times>, "\001\000\000\000\000\000\000\000�\215\213", '\0' <repeats 13 times>, "\004", '\0' <repeats 15 times>, "\b\236\005I�\177\000\000\227%J\000\000\000\000" #7 0x0000000000467873 in main (argc=<value optimized out>, argv=0x7fff49059e08) at main.c:2398 cfg_stream = (FILE *) 0x12e1010 c = <value optimized out> r = <value optimized out> tmp = 0x7fff4905ae90 "" tmp_len = 32520 port = <value optimized out> proto = <value optimized out> ret = <value optimized out> seed = 1235801225 ---Type <return> to continue, or q <return> to quit--- rfd = 4 debug_save = <value optimized out> debug_flag = 0 dont_fork_cnt = 0 n_lst = <value optimized out> p = <value optimized out> (gdb) (gdb) quit kamailio2:/var/kamailio# kamailio -V version: kamailio 3.1.2 (x86_64/linux) eb24c1-dirty flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: eb24c1 -dirty compiled on 09:35:52 Apr 28 2011 with gcc 4.3.2

2 5

geoip problem, finally solved
by Francesco Castellano 16 May '11

16 May '11

Dear Sirs, I found some problems using the geoip module. My system is a debian squeeze x86_64; and I installed the following packages: Desired=Unknown/Install/Remove/Purge/Hold | Status=Not/Inst/Conf-files/Unpacked/halF-conf/Half-inst/trig-aWait/Trig-pend |/ Err?=(none)/Reinst-required (Status,Err: uppercase=bad) ||/ Nome Versione Descrizione +++-=============================================-=============================================-========================================================================================================== ii geoip-bin 1.4.7~beta6+dfsg-1 IP lookup command line tools that use the GeoIP library ii geoip-database 1.4.7~beta6+dfsg-1 IP lookup command line tools that use the GeoIP library (country database) ii libgeoip-dev 1.4.7~beta6+dfsg-1 Development files for the GeoIP library ii libgeoip1 1.4.7~beta6+dfsg-1 A non-DNS IP-to-country resolver library Using /usr/share/GeoIP/GeoIP.dat as database, the geoip_match("$si", "src") function failed with retcode -2 (NULL from GeoIP_record_by_name()). And finally I found the problem with the GeoIP file DB itself. In particular, compiling a mini program for debug, it said: Invalid database type GeoIP Country Edition, expected GeoIP City Edition, Rev 1 After manually downloading the GeoLiteCity.dat from MaxMind.com everything went fine. I wrote for letting others know my solution, and for asking you if there is a cleaner way to solve it. Best regards, Francesco Castellano

2 3

nathelpler module path support
by Asgaroth 16 May '11

16 May '11

Hi All, I have a scenario where our registrar's are sitting behind our sip proxy. The sip proxy is the only system that has access to the outside world. Our registrar's are all on internal private addresses. I am trying to configure our registrar's to perform nat pinging. However, because the registrar's are on an internal address there is no way for the nat ping to get out. I enabled the path option so that the registrar stores the path that the registration comes in on. However it doesnt seem like the nathelper module uses the path parameter to send the outgoing nat ping. I tried the nat ping using sip method INFO, but the same applies, it does not appear to be able to send out via the outbound proxy. I tried to enable the natping option on the sip proxy, but it appears that only registrations with a socket address of the local host will get pinged, or else it is ignored. Is there an option that I am missing for the nat helper module that will enable me to set an outbound proxy for the nat ping messages? Any insight would be appreciated. Thanks

3 6

LCR strip and tags (prefix) - Kamailio 3.1
by Alexandre Abreu 14 May '11

14 May '11

Hi, With this LCR release from Kamailio 3.1 the table 'lcr_gw' makes four fields unique: lcr_id, ip_addr, port and hostname. This table is also responsible for setting 'strip' and 'tag' parameters values for the gateways. What I am trying to accomplish is to use different strip and tag values for the SAME gateway based only on what was dialed by the user. In 1.5 LCR for example, if user dialed 444 or 555 and both prefix belongs to the same gateway, I could use gw-sip for 444 (with values strip 1 and prefix 22) and also gw-sip for 555 (with values strip 2 and prefix 33). While this duplicate data, it was possible to play with routes without touching the config file. I am looking for ideas on how to implement that on K3.1 without using ifs, strip and prefix tied (statically) in the script file. As far as I know, for every new lcr_id inserted, I will need a corresponding load_gws(new_lcr_id) in the script. Regards, Alexandre.

2 1

DBTEXT - authentication against text files
by Pavel Segeč 13 May '11

13 May '11

Hi, does anybody have some knowledge or experience with DBTEXT module? I'm searching for working DBTEXT files examples and config files examples... thanks palo

1 0

[OT] SIP presence and BLF testing tools
by Andrew Pogrebennyk 12 May '11

12 May '11

Is there any framework available to do functional and regression testing of SIP presence and BLF? Ideally I'm looking for a tool for asserting the signaling flow and contents of the XML body (state, version numbers etc), also I'd like to be able to extract the tags and call-id into variables so as to create INVITE with Replaces. Working with message body variables in sipp is a pain. I'm also familiar with SIPr, Net::SIP and a few others, but that doesn't look suitable for someone without much development experience. Spirent is a perhaps a little closer than others in implementing pickup scenario but awkward when you go beyond the default test scenarios. So I'm interested in any information, tips, suggestions, commercial tools etc. I'm asking here 'cause judging from the quality of their BLF&presence OpenSIPS and Kamailio got the testing process just right. -- Sincerely, Andrew Pogrebennyk

2 1

core dump
by Dominguez Jover, Ricardo 12 May '11

12 May '11

Hi everybody, We are having an uncontrolled crash in Kamailio 3.1: May 10 20:34:51 tip1 /usr/local/kamailio-3.1/sbin/kamailio[23366]: : <core> [pass_fd.c:293]: ERROR: receive_fd: EOF on 39 May 10 20:34:51 tip1 /usr/local/kamailio-3.1/sbin/kamailio[23327]: ALERT: <core> [main.c:741]: child process 23364 exited by a signal 11 May 10 20:34:51 tip1 /usr/local/kamailio-3.1/sbin/kamailio[23327]: ALERT: <core> [main.c:744]: core was not generated I've read in several posts I must generate the "Core Dump" to know what is happening. I'm trying to generate it in this way. "/etc/init.d/Kamailio" file: if test "$DUMP_CORE" = "yes" ; then 1. set proper ulimit ulimit -c unlimited directory for the core dump files COREDIR=/dumps/ [ -d $COREDIR ] || mkdir $COREDIR chmod 777 $COREDIR echo "$COREDIR/core.%e.sig%s.%p" > /proc/sys/kernel/core_pattern fi "/etc/default/Kamailio" file: DUMP_CORE=yes But the core is not yet generated. Then I've added -w option: OPTIONS="-P $PID_FILE -m $MEMORY -u $USER -g $GROUP -w /dumps/" But core still not generated In "kamailio.cfg" I´ve also added: disable_core_dump=no User running Kamailio is "kamailio" who has 777 permissions in /dumps/ directory. No way. Anyhelp would be appreciated. Cheers, Ricardo Domínguez

3 5

Monitor SIP Trunk Usage with MRTG
by JR Richardson 11 May '11

11 May '11

Hi All, So I'm still banging away at monitoring various SIP trunk capacity, the dialog module is working well for this purpose. I am running local server cron job scripts to pull data and put into local flat files for reference. My next step is to pull this data from a remote server and graph with MRTG (this is what I use for most other network monitoring). It looks like I can use the mi_xmlrpc module and remotely query the SIP Router with kamctl commands and parse the results into MRTG. I'm interested in hearing from others that may have already done this and would not mind sharing some scripts or from anyone that has an alternative method. Any guidance or affirmation that this is a good method that works would be helpful. On the MRTG server, I'm using perl scripts for a lot of my data pulling. I am using Kamailio 3.0. Thanks. JR -- JR Richardson Engineering for the Masses

2 3

Kamailio + RTP Proxy modification -- programmer needed
by Salauddin Rubel 11 May '11

11 May '11

Hi, We need a programmer who can modify SIP and RTP packets passing through Kamailio and RTP proxy. Interested programmer can contact with me. Thanks

3 3

Dialog Module Question inbound/outbound count
by JR Richardson 10 May '11

10 May '11

Hi All, Been playing around with the dialog module to help me keep track of SIP trunk utilization, trying to identify how many calls are active on each SIP trunk. My dialog count for inbound calls is 10 (correct) but my outbound calls are always double that and I'm not sure why. Here is a snip of my config: loadmodule "dialog.so" if($rU=~"^111") { dlg_manage(); dlg_setflag("4"); set_dlg_profile("inbound","trunk111"); route(PDT); exit; } PDT lookup for prefix then call is forwarded to dispatcher to route out to carrier through 'route[relay]' route[RELAY] { dlg_setflag("4"); set_dlg_profile("outbound","$rd"); if (!t_relay()) { sl_reply_error(); exit; } exit; } I setup 10 active calls and here are my results: test-sip-router2:~# kamctl fifo profile_get_size inbound profile:: name=inbound value= count=10 test-sip-router2:~# kamctl fifo profile_get_size outbound profile:: name=outbound value= count=20 Also 'kamctl fifo dlg_list' will list only 10 active dialogs. So my question is, why am I getting a 20 count for outbound and only 10 count for inbound? Thanks. JR -- JR Richardson Engineering for the Masses

2 2

t_reply MI Datagram Function
by Torge Szczepanek 10 May '11

10 May '11

Hi! In the Documentation of the TMX Module for the t_reply function there is given a list of Parameters as can be seen here: http://www.kamailio.org/docs/modules/3.1.x/modules_k/tmx.html#id2822345 The third parameter is a trans_id. I cannot find any references for this. Is it possible to send a query to the MI Datagram socket, to do a lookup of the trans_id for a given callid or is there any other solution to get this information through the MI datagram socket, to be able to execute t_reply? It seems that t_reply_callid is doing what I want to do, but currently not through the MI Datagram socket and not including a body or new headers. Greetings Torge

2 2

INSTALL file
by Claudio Furrer 10 May '11

10 May '11

Hello, Is this still applicable? See INSTALL file (line 669) 3.1) create MySQL tables for SER flavour - if you have a previously installed SER on your system, use /usr/sbin/ser_mysql.sh reinstall to convert your SER database into new structures - otherwise, if this is your very first installation, use /usr/sbin/ser_mysql.sh create to create SER database structures (you will be prompted for password of MySql "root" user) Because I did not found the *reinstall* option on sources.. Thanks, Claudio

2 1

Re: [SR-Users] SIP Trunk Usage Monitoring
by JR Richardson 10 May '11

10 May '11

> On 09.05.2011 22:44, JR Richardson wrote: > > Would there be any usage examples of dialog module? I'm not sure if I > > really need profiles and [values] and when to set and unset. A push > > in the right direction would be helpful. > > Hi, > here is an example I've used for incoming and outgoing calls tracking, > just tune it for your purposes: > > loadmodule "dialog.so" > modparam("dialog", "dlg_flag", 4) > modparam("dialog", "db_mode", 1) > modparam("dialog", "profiles_with_value", "carrierin;carrierout") > ... > create_dialog(); > if(is_from_gw()) > { > # Monitor number of incoming calls > get_profile_size("carrierin","$si","$avp(s:cnt)"); > xlog("L_INFO", "----- currently, the carrier $si has > $avp(s:cnt) active outgoing calls\n"); > set_dlg_profile("carrierin","$si"); > } else { > # Monitor number of outgoing calls > get_profile_size("carrierout","$rd","$avp(s:cnt)"); > xlog("L_INFO", "----- currently, the carrier $rd has > $avp(s:cnt) active outgoing calls\n"); > set_dlg_profile("carrierout","$rd"); > } > > -- > Sincerely, > Andrew Pogrebennyk Thanks Andrew, very nice example, this will definitely help. Have a great day. JR

1 0

Re: [SR-Users] Dialog Module Usage Question
by JR Richardson 10 May '11

10 May '11

> > The problem I am having specifically is for outbound calls flowing > > through a dispatcher, I want to separate calls into a profile for each > > SIP domain that I send calls to. I'm not sure how do accomplish this. > > I see in kamctl fifo dlg_list, there is 'caller_bind_addr:: > > udp:10.10.12.24:5060'. Is there a way to export that into a variable > > that I can use as the 'profile' for the dialog? > > The domain part of the request URI is stored in a pseudovariable > called $rd. I would just use that as your profile 'value'. > > -- > Alex Balashov - Principal Thanks Alex, after I walked away from it for a bit, that came to mind. Really appreciate the feedback. Thanks. JR

1 0

Re: [SR-Users] SIP Trunk Usage Monitoring
by JR Richardson 10 May '11

10 May '11

>> I recently interconnected to an upstream carrier using Kamailio 3.0, >> working fine. We have configure 2 SIP trunks for failover/redundnacy. >> I'm using dispatcher module to round robin calls to the carrier. I >> wanted to monitor trunk usage between us. I was reading in the devel >> 3.2 about some native monitoring in the new dispatcher module without >> the need to load the dialog module. >> >> Can I use the new module in version 3.0 or will I need to upgrade the >> core to 3.2? >> >> I typically will not use dev software in production, so can anyone >> suggest a light method of trunk usage monitoring for ver 3.0? I'm not >> opposed to using the dialog module, I'm just not familiar with it and >> not sure the best way to integrate it for usage tracking. I don't >> need to do billing or capture cdr's on the SIP-Routers so the simpler >> the better. > > Honestly, going the dialog module route is your easiest bet. > > -- > Alex Balashov - Principal Thanks Alex, Would there be any usage examples of dialog module? I'm not sure if I really need profiles and [values] and when to set and unset. A push in the right direction would be helpful. Thanks. JR -- JR Richardson Engineering for the Masses

2 1

Dialog Module Usage Question
by JR Richardson 10 May '11

10 May '11

Hi All, I'm trying to configure the dialog module with profiles for outbound calls to different Dispatcher domains. I can get the dialog module configured for inbound calls and pull statistics and separate calls into specific profiles, that part is working well. The problem I am having specifically is for outbound calls flowing through a dispatcher, I want to separate calls into a profile for each SIP domain that I send calls to. I'm not sure how do accomplish this. I see in kamctl fifo dlg_list, there is 'caller_bind_addr:: udp:10.10.12.24:5060'. Is there a way to export that into a variable that I can use as the 'profile' for the dialog? Thanks. JR -- JR Richardson Engineering for the Masses

2 2

Ser AVP
by Claudio Furrer 09 May '11

09 May '11

Hello all, Have this issue with my ser config. I try to load a fwdbusy value from db using the $fu class, but it's complaining.. config: ... if (avp_db_load("$fu", "s:fwdbusy")) { ... log: 0(20749) ERROR: <core> [pvapi.c:516]: error searching pvar "fu" 0(20749) ERROR: <core> [pvapi.c:720]: wrong char [u/117] in [$fu] at [2 (0)] 0(20749) ERROR: avpops [avpops.c:275]: bad param 1; expected : $pseudo-variable or int/str value What's wrong with that? Should it not considered as a ser avp? Not K one. Or that avp_db_load is an exlusively K function? Thank you, Claudio

2 2

Kamailio fails to register to a VoIP provider
by Pierpaolo Culurciello 09 May '11

09 May '11

Dear all, I'm trying to register Kamailio (version 3.1.3) to different VoIP provider but I always receive a 401 unauthorized even if I use valid credentials. I post below two different capture of the registration to the same VoIP provider using the same credentials (SIP domain 94.23.56.123, SIP auth username 1028), with a Blink softphone (successful) and with Kamailio 3.1.3 (denied). The public IP address of my Kamailio is anonymized with XXXXXXXX: ------ Blink softphone -------- REGISTER sip:94.23.56.123 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.1:49186;rport;branch=z9hG4bKPjGtybPeuocgJbvOZlRNoWG3RPL0peCs31 Max-Forwards: 70 From: "ItaliADSL" <sip:1028@94.23.56.123>;tag=Tid9EUBxo3Bc5Eo0xzaMg82Vk8q7J57j To: "ItaliADSL" <sip:1028@94.23.56.123> Contact: <sip:alybrkhq@192.168.2.115:49186> Call-ID: cdr3Rr05eEs0voSSZ275zLxVF.Eo74mo CSeq: 1 REGISTER Expires: 600 User-Agent: Blink 0.22.2 (MacOSX) Content-Length: 0 SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.2.1:49186;branch=z9hG4bKPjGtybPeuocgJbvOZlRNoWG3RPL0peCs31;received=95.248.232.55;rport=49186 From: "ItaliADSL" <sip:1028@94.23.56.123>;tag=Tid9EUBxo3Bc5Eo0xzaMg82Vk8q7J57j To: "ItaliADSL" <sip:1028@94.23.56.123>;tag=as5a82d867 Call-ID: cdr3Rr05eEs0voSSZ275zLxVF.Eo74mo CSeq: 1 REGISTER User-Agent: MOR Softswitch Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6afe07ca" Content-Length: 0 REGISTER sip:94.23.56.123 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.1:49186;rport;branch=z9hG4bKPj.JA-9Zf0dpKV6v0JmKtOGSMhyNw5ep1w Max-Forwards: 70 From: "ItaliADSL" <sip:1028@94.23.56.123>;tag=Tid9EUBxo3Bc5Eo0xzaMg82Vk8q7J57j To: "ItaliADSL" <sip:1028@94.23.56.123> Contact: <sip:alybrkhq@192.168.2.115:49186> Call-ID: cdr3Rr05eEs0voSSZ275zLxVF.Eo74mo CSeq: 2 REGISTER Expires: 600 User-Agent: Blink 0.22.2 (MacOSX) Authorization: Digest username="1028", realm="asterisk", nonce="6afe07ca", uri="sip:94.23.56.123", response="f3489b3c8510f39bcf3d1fd96b8836fb", algorithm=MD5 Content-Length: 0 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.1:49186;branch=z9hG4bKPj.JA-9Zf0dpKV6v0JmKtOGSMhyNw5ep1w;received=95.248.232.55;rport=49186 From: "ItaliADSL" <sip:1028@94.23.56.123>;tag=Tid9EUBxo3Bc5Eo0xzaMg82Vk8q7J57j To: "ItaliADSL" <sip:1028@94.23.56.123>;tag=as5a82d867 Call-ID: cdr3Rr05eEs0voSSZ275zLxVF.Eo74mo CSeq: 2 REGISTER User-Agent: MOR Softswitch Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Expires: 600 Contact: <sip:alybrkhq@192.168.2.115:49186>;expires=600 Date: Sun, 08 May 2011 20:16:58 GMT Content-Length: 0 ----------- Kamailio 3.1.3 ---------------- REGISTER sip:94.23.56.123 SIP/2.0 Via: SIP/2.0/UDP XXXXXXXXXXX;branch=z9hG4bK58b4.3ac3b834.0 To: sip:1028@94.23.56.123 From: sip:1028@94.23.56.123;tag=464474ecb420f5357c1361be147933f0-45fd CSeq: 10 REGISTER Call-ID: 6efffe60-27704@XXXXXXXXXXXX Content-Length: 0 User-Agent: kamailio (3.1.3 (i386/linux)) Contact: <sip:1028@XXXXXXXXXX> Expires: 360 SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP XXXXXXXXXX;branch=z9hG4bK58b4.3ac3b834.0;received=XXXXXXXXXX From: sip:1028@94.23.56.123;tag=464474ecb420f5357c1361be147933f0-45fd To: sip:1028@94.23.56.123;tag=as322a1728 Call-ID: 6efffe60-27704@XXXXXXXXXXX CSeq: 10 REGISTER User-Agent: MOR Softswitch Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5f3ee3e8" Content-Length: 0 REGISTER sip:94.23.56.123 SIP/2.0 Via: SIP/2.0/UDP XXXXXXXXXX;branch=z9hG4bK7666.c1b2e93.0 To: sip:1028@94.23.56.123 From: sip:1028@94.23.56.123;tag=464474ecb420f5357c1361be147933f0-1f9c CSeq: 10 REGISTER Call-ID: 6efffe60-27697@XXXXXXXXXXX Content-Length: 0 User-Agent: kamailio (3.1.3 (i386/linux)) Contact: <sip:1028@XXXXXXXXXXX> Expires: 360 Authorization: Digest username="1028", realm="asterisk", nonce="5f3ee3e8", uri="sip:94.23.56.123", response="df6ebfa9b5870b8b912d1e3b53eedc62", algorithm=MD5 SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP XXXXXXXXX;branch=z9hG4bK7666.c1b2e93.0;received=XXXXXXXXX From: sip:1028@94.23.56.123;tag=464474ecb420f5357c1361be147933f0-1f9c To: sip:1028@94.23.56.123;tag=as529fc11d Call-ID: 6efffe60-27697@XXXXXXXXXX CSeq: 10 REGISTER User-Agent: MOR Softswitch Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="41f3fe77" Content-Length: 0 -------------------------------------- As you can see from the captures, Blink softphone successfully registrates with VoIP provider while Kamailio's request is rejected. I tried to analize the two flows of SIP messages but I didn't notice any difference between them so I can't understand why Kamailio is not able to register. Can anybody help me to solve this issue? greetings, Pierpaolo

2 1

Authenticate if receiving 401
by Eric Hiller 09 May '11

09 May '11

I want kamailio to authenticate itself to a host if it is sent a 401, just as that host is expected to authenticate if kamailio sends it one. I am not finding much in the online probably because I am not searching for the right terms. Does anyone have any experience in this? Thanks! -Eric

4 4

Kamailio BLF - NOT SIMPLE
by Lucas Alvarez 09 May '11

09 May '11

Hi, I've been trying to configure kamailio for enabling BLF for Aastra phones. Sincerely, it's been a nightmare, till now I'm not sure if I have to enable pua_dialoginfo or presence_dialoginfo module or both for handling dialog events. Besides that is not clear which changes should be made to the route configuration nor which parameters should be configured. I would appreciate if anyone can enlighten me or could point me to the right documentation page. Thanks in advance. Lucas Alvarez

2 1

Force RTPProxy
by Simone Felici 09 May '11

09 May '11

Hello to all! I need a little help with our ser installation (ser-2.0.0-rc1). Due some network restrictions, we need to force every call to pass trought rtpproxy. Could someone point me to the right place where to force it? Please an example based on our config would be really helpful, but in two scenarios: 1. ALL CALLS trought rtpproxy 2. ONLY CALLS from a given network trought rtpproxy We are in doubt on what choose between the two solutions. Having both examples could me help to implement the needs faster :) Thank's a lot for your help! Simon

2 5

rewriteuri and append_branch with variables
by alex pappas 07 May '11

07 May '11

Dear All, I'm trying the following and i get error: $var(new_uri) = "sip:" +$avp(s:term_prefix) +$rU +"@" +$avp(s:gw_address); rewriteuri=($var(new_uri)); OR append_branch($var(new_uri)); But it seems that it does not like the variable. The documentation says that rewriteuri get a string so i don't understand why does not work. Do you have any ideas on this ? Cheers Alex

3 4

SIP Trunk Usage Monitoring
by JR Richardson 06 May '11

06 May '11

Hi All, I recently interconnected to an upstream carrier using Kamailio 3.0, working fine. We have configure 2 SIP trunks for failover/redundnacy. I'm using dispatcher module to round robin calls to the carrier. I wanted to monitor trunk usage between us. I was reading in the devel 3.2 about some native monitoring in the new dispatcher module without the need to load the dialog module. Can I use the new module in version 3.0 or will I need to upgrade the core to 3.2? I typically will not use dev software in production, so can anyone suggest a light method of trunk usage monitoring for ver 3.0? I'm not opposed to using the dialog module, I'm just not familiar with it and not sure the best way to integrate it for usage tracking. I don't need to do billing or capture cdr's on the SIP-Routers so the simpler the better. Any guidance will be much appreciated. Thanks. JR -- JR Richardson Engineering for the Masses

2 1

t_check_status documentation
by Eric Hiller 06 May '11

06 May '11

One thing that really threw me this morning was the lack of t_check_status in the modules documentation. Apparently it is still there and works, just completely undocumented. Once I realized it was still there I just went back to the 1.5 docs and was good, but just thought I should point this out. -Eric

2 1

db_text issue
by Eric Hiller 06 May '11

06 May '11

I would like to use dbtext to store a list of whitelisted gateways, but each time I load it I get a core segmentation fault. Any ideas? Here is my gateways db file: key_name(str) key_type(int) value_type(int) key_value(str) gwip:1:0:a gwip:1:0:b gwip:1:0:c gwcount:0:1:3 And in my config: modparam("htable","htable","gateways=>size=4;dbtable=gateways;") modparam("htable","db_url","text:///usr/local/etc/kamailio") while( $var(i) < $sht(gateways=>gwcount) ){ if( $sht(gateways=>gwip[$var(i)]) == $si ){ setflag(gateway); xlog("ISFROMGW: request from IP $si matches $sht(gateways=>gwip[$var(i)]) ...\n"); } } Do you see anything wrong with this setup? -Eric

2 4

free tickets linuxtag
by Henning Westerholt 06 May '11

06 May '11

Hello, we've few free 4 day tickets for the LinuxTag next week in Berlin to share. If you like to receive one, just send me a short reply. More informations about our booth: http://www.kamailio.org/w/2011/04/kamailio-booth-at-linuxtag-2011/ Best regards, Henning

1 0

Force RTPProxy
by Simone Felici 06 May '11

06 May '11

Hello to all! I need a little help with our ser installation (ser-2.0.0-rc1). Due some network restrictions, we need to force every call to pass trought rtpproxy. Could someone point me to the right place where to force it? Please an example based on our config would be really helpful, but in two scenarios: 1. ALL CALLS trought rtpproxy 2. ONLY CALLS from a given network trought rtpproxy We are in doubt on what choose between the two solutions. Having both examples could me help to implement the needs faster :) Thank's a lot for your help! Simon

1 0

Calling t_on_reply in branch_route doesn't work in Kamailio 3.1 - did work in Kamailio 1.5
by Sebastian Damm 05 May '11

05 May '11

Hi, I'm currently migrating a Kamailio 1.5 installation to Kamailio 3.1.3, and I got all config changes merged, except this one: branch_route[1] { if ((isbflagset(4)) && (!isflagset(7))) { fix_nated_sdp("1"); if (has_body()) { force_rtp_proxy("R"); } else { setflag(5); } t_on_reply("2"); } } In Kamailio 1.5 this worked without problems. In Kamailio 3.1 I get the following error (line 1627 is the line with t_on_reply on it): hostname:~# /usr/sbin/kamailio -c -f /etc/kamailio/proxy/kamailio.cfg loading modules under /urs/lib/kamailio/modules/:/usr/lib/kamailio/modules_k:/usr/lib/kamailio/modules_s/ 0(13670) : <core> [cfg.y:3412]: parse error in config file /etc/kamailio/proxy/kamailio.cfg, line 1627, column 17: Command cannot be used in the block ERROR: bad config file (1 errors) hostname:~# In the Kamailio 1.5 documentation it was stated for every t_on_* function, where it could be called. For t_on_reply it said: This function can be used from REQUEST_ROUTE, BRANCH_ROUTE, ONREPLY_ROUTE and FAILURE_ROUTE. In the Kamailio 3.1 documentation those lines are missing. But I don't really see a reason, why t_on_reply shouldn't be called from branch route. We need it for every branch to possibly have different onreply_routes. Am I missing something? Thanks in advance for any hints. Best regards, Sebastian

3 6

Force RTPProxy
by Simone Felici 05 May '11

05 May '11

Hello to all! I need a little help with our ser installation (ser-2.0.0-rc1). Due some network restrictions, we need to force every call to pass trought rtpproxy. Could someone point me to the right place where to force it? Please an example based on our config would be really helpful, but in two scenarios: 1. ALL CALLS trought rtpproxy 2. ONLY CALLS from a given network trought rtpproxy We are in doubt on what choose between the two solutions. Having both examples could me help to implement the needs faster :) Thank's a lot for your help! Simon

1 0

Re: [SR-Users] Kamailio lib/srdb1/db_id.c handling of multiple database connections and race condition
by Robert Verspuy 05 May '11

05 May '11

Hi Torge, I think it's better to send your mail to the mailinglist, then more people can have a look at your problem. I'm not really familiar with the kamailio source code. Just started using Kamailio about 8 months back. I came across a problem with requesting information from a postgresql database and made a patch which 'works for me'. I'm not using siptrace. Maybe you can supply more logging (through the mailinglist) of the database communication. The mysql error "Duplicate entry '96193' for key 'PRIMARY'" is a bit weird. As I said, I don't know and don't use the msyql driver in kamailio, but normally I would expect that mysql itself would take care of generating a primary key for the record. So maybe your problem is in the mysql server and not kamailio. Maybe this page can help you: http://www.softwareprojects.com/resources/programming/t-how-to-fix-mysql-du… With kind regards, Robert Verspuy Op 05-05-11 13:19, Dr. Torge Szczepanek schreef: > Hi! > > First of all thanks for all of your work. I really appreciate and I am > glad that there is such a nice Opensource SIP Proxy available. > > In a current project I am heavily relying on SIP Traces stored to a > database and other sqlops, dialplan operations using database connections. > > It seems that there is a race condition with SQL Databases (Postgres > and also MySQL) due to thread insafety as described in: > > http://lists.sip-router.org/pipermail/sr-dev/2010-October/009689.html > > This is causing multiple errors. I guess that the following error is > also caused by this race condition (thread safety): > > /usr/sbin/kamailio[5646]: ERROR: <core> [db_query.c:179]: error while > submitting query > /usr/sbin/kamailio[5646]: ERROR: siptrace [siptrace.c:532]: error > storing trace > /usr/sbin/kamailio[5646]: ERROR: db_mysql [km_dbase.c:120]: driver > error on query: Duplicate entry '96193' for key 'PRIMARY' > > My Kamailo Version (3.1.2) is reporting duplicate database URLs using > a warning: > > /usr/sbin/kamailio[5613]: WARNING: <core> [db_id.c:281]: identical DB > URLs, but different DB connection pid [5613/5562] > /usr/sbin/kamailio[5618]: WARNING: <core> [db_id.c:281]: identical DB > URLs, but different DB connection pid [5618/5562] > /usr/sbin/kamailio[5616]: WARNING: <core> [db_id.c:281]: identical DB > URLs, but different DB connection pid [5616/5562] > /usr/sbin/kamailio[5600]: WARNING: <core> [db_id.c:281]: identical DB > URLs, but different DB connection pid [5600/5562] > /usr/sbin/kamailio[5617]: WARNING: <core> [db_id.c:281]: identical DB > URLs, but different DB connection pid [5617/5562] > > Git commit 9fb37a55e0b97a98b7d2d104a7692f18cb3f7df6 is turning this > into a debug message, which I think is a bad idea if the thread > insafety still exists. > > As far as I know every thread should use its own database handle, at > least when using the MySQL API. > > Is there a way to handle this correctly? Can you give me any clues on > which approach to take here? Since I need a solution for this I would > like to fix this issue at least for myself and of course giving the > change back. > > Greetings Torge > -- > Dr. Torge Szczepanek > > Cygnus Networks GmbH Tel: +49-228-9090987-0 > Adenauerallee 73 Fax: +49-228-9090987-70 > D-53113 Bonn EMail: > info(a)cygnusnetworks.de <mailto:info@cygnusnetworks.de> > Amtsgericht Bonn, HRB 15928 SIP: info(a)cygnusnetworks.de > <mailto:info@cygnusnetworks.de> > Geschaeftsfuehrer Dr. Torge Szczepanek http://www.cygnusnetworks.de/ > Steuernummer 205/5710/1347 USt-ID DE258507007 > -- *Exa-Omicron* Patroonsweg 10 3892 DB Zeewolde Tel.: 088-OMICRON (66 427 66) http://www.exa-omicron.nl

2 1

lcr module: strip and tag
by Alida - Danilo SMALDONE 05 May '11

05 May '11

Hello, I've some understanding problem on lcr module. I've this route route[LCR] { xlog("L_INFO", "ENTRO IN ROUTE LCR ********************************** \n"); if (!load_gws(1)) { sl_send_reply("503", "Unable to load gateways"); exit; } else { $var(i) = 0; while(is_avp_set("$(avp(i:709)[$var(i)])")) { xlog("L_INFO","loading gw_uri_avp[$var(i)]='$(avp(i:709)[$var(i)])'\n"); $var(i) = $var(i) + 1; avp_print(); }; if(is_avp_set("$avp(i:709)")) { xlog("L_INFO", "Trying gateway '$avp(i:709)'\n"); } else { xlog("L_INFO", "No More Gateways \n"); }; # try the first matched gateway if (next_gw()) { xlog("L_INFO","ruri_user_avp='$avp(i:500)'\n"); xlog("L_INFO","test ruri_avp_mod='$tu'\n"); #xlog("L_INFO","************************************************************ ****\n"); avp_print(); # Route to failure for failover #t_on_failure("2"); route(RELAY); } else { sl_send_reply("503","No available gateways"); exit; }; }; exit; } And this is what I get: Apr 27 15:44:44 localhost kamailio[7827]: INFO: <script>: ENTRO IN ROUTE LCR ********************************** Apr 27 15:44:44 localhost kamailio[7827]: INFO: <script>: loading gw_uri_avp[0]='1|1|3|774|1171521237|172.16.10.200|5060||1|0' Apr 27 15:44:44 localhost kamailio[7827]: INFO: avpops [avpops_impl.c:1328]: p=0xb60dc5b0, flags=0x0112 Apr 27 15:44:44 localhost kamailio[7827]: INFO: avpops [avpops_impl.c:1334]: id=<709> Apr 27 15:44:44 localhost kamailio[7827]: INFO: avpops [avpops_impl.c:1340]: val_str=<1|1|3|774|1171521237|172.16.10.200|5060||1|0 / 45> Apr 27 15:44:44 localhost kamailio[7827]: INFO: <script>: Trying gateway '1|1|3|774|1171521237|172.16.10.200|5060||1|0' Apr 27 15:44:44 localhost kamailio[7827]: INFO: <script>: ruri_user_avp='09711234567' Apr 27 15:44:44 localhost kamailio[7827]: INFO: <script>: test ruri_avp_mod='sip: 09711234567(a)voip.myproxy.it' Apr 27 15:44:44 localhost kamailio[7827]: INFO: avpops [avpops_impl.c:1328]: p=0xb60dc624, flags=0x0110 Apr 27 15:44:44 localhost kamailio[7827]: INFO: avpops [avpops_impl.c:1334]: id=<712> Apr 27 15:44:44 localhost kamailio[7827]: INFO: avpops [avpops_impl.c:1342]: val_int=<0> Apr 27 15:44:44 localhost kamailio[7827]: INFO: avpops [avpops_impl.c:1328]: p=0xb60dc5b0, flags=0x0112 Apr 27 15:44:44 localhost kamailio[7827]: INFO: avpops [avpops_impl.c:1334]: id=<500> Apr 27 15:44:44 localhost kamailio[7827]: INFO: avpops [avpops_impl.c:1340]: val_str=<09711234567 / 11> I expected that the lcr module strips 3 numbers from avp(i:500) and adds the prefix 774. Anyone can explane why it didn't works? The called number never changes in 77409711891234567. Any help will be appreciated. Thank you Danilo

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Rtp Port Question
by Sungjin Park 04 May '11

04 May '11

Hello, I use your sip service well. But there is one problem occured. I use your sip service on my android system. The application is Csipsimple. On Wifi and 3G connection, your service is very good. your sip account did good work. even 3G network connection. Btw, On Mobile Wimax service, the signal was good. But the Voice could not be trasnsferred. I guess the problem is about rtp port. *I'd like to know the rtp port that IPTEL use*. I will request Korea Telecom about rtp port open. Korea Telecom is Korean mobile Wimax Provider. Thank you. Sungjin Park Seoul, Korea

2 1

Re: [SR-Users] ? about presence
by hb 04 May '11

04 May '11

Thank you for your attention. I use the soft phone "jitsi" as is told in guide file and configure it as the guide. And I have checked the guide twice. I think there have nothing different with the guide excepts the db I used. >From: Martin Pohan?en?k <pohancenik.martin(a)gmail.com> >Subject: Re: [SR-Users] ? about presence >To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List" <sr-users(a)lists.sip-router.org> >Message-ID: <BANLkTi=5S3OHe2SEDDe8ry+X+2ZUd2LqRg(a)mail.gmail.com> >Content-Type: text/plain; charset="gb2312" > >What SIP client app are you using to test presence ? >Please try to post the client configuration, too. > >Also try to go through the entire config file from the guide and compare >what is different from your config file. That might give you a clue of what >is wrong > >2011/5/3 Bucur Marius <bucur_marius_ovidiu(a)yahoo.com> > >> Hello, >> >> If you disable "#!define WITH_PRECENCE" you will not use presence at all - >> so it is most probably that you don't want that. >> When you use db_fallback mode, the subscribers are only written to database >> once in a while (in your case every 20 seconds). >> Also, if the subscribers are not authorized, they will only be in the >> "watchers" table - not active_watchers. >> Can you check if there is something in the "watchers" table? >> Also, can you check your system and see if it works without the xcap >> server? >> >> modparam("presence_xml", "force_active", 1) >> #modparam("presence_xml", "integrated_xcap_server", 1) >> >> Cheers, >> Marius >> >> ------------------------------ >> *From:* hb <hhb.1988(a)163.com> >> *To:* sr-users(a)lists.sip-router.org >> *Sent:* Tue, May 3, 2011 11:44:16 AM >> *Subject:* [SR-Users] ? about presence >> >> Dear everyone: >> I use kamalio 3.1. >> I want to add the xcap_sever and presence modules into my >> kamailio. I followed the lists " >> http://cache.baidu.com/c?m=9d78d513d99a12f00bb6ca255001d7175b14d7226b8b8142…" >> step by step. The only difference is that i use the database postgresql. >> When i start the server, it worked well, but the softphone was not >> able to show the presence of the contacts. >> I tried many times and found that when i disable the sentence >> "#!define WITH_PRECENCE" in "kamailio.cfg", it can show the state of the >> contacts. However, when i check the database, there was nothing in "Active >> Watchers" and "Presentity" tables. >> I think there must have something wrong with the configure of >> presence. Any response will be appreciated. >> >> >> >> ps: the configuration of the presence in kamailio.cfg as follows(same as >> the Tutorial): >> >> #!ifdef WITH_PRESENCE >> loadmodule "presence.so" >> loadmodule "presence_xml.so" >> #!endif >> >> #!ifdef WITH_PRESENCE >> # ----- presence params ----- >> modparam("presence", "db_url", DBURL) >> modparam("presence", "fallback2db", 1) >> modparam("presence", "db_update_period", 20) >> >> # ----- presence_xml params ----- >> modparam("presence_xml", "db_url", DBURL) >> modparam("presence_xml", "force_active", 0) >> modparam("presence_xml", "integrated_xcap_server", 1) >> #!endif >> haibo huang >> >> >> >> ------------------------------ >> ??????2G????????????????????3?!<http://mail.163.com/html/110414_attachment/att1.htm> >> >> _______________________________________________ >> 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 >> >> >-------------- next part -------------- >An HTML attachment was scrubbed... >URL: <http://lists.sip-router.org/pipermail/sr-users/attachments/20110503/8d425fb…> > >------------------------------ > >Message: 2 >Date: Tue, 3 May 2011 12:55:57 +0900 >From: Sungjin Park <parksj84(a)gmail.com> >Subject: [SR-Users] Rtp Port Question >To: serusers(a)iptel.org >Message-ID: <BANLkTimJwR5e_yM5SA67QjYzOj8s=tgAjQ(a)mail.gmail.com> >Content-Type: text/plain; charset="iso-8859-1" > >Hello, I use your sip service well. >But there is one problem occured. > >I use your sip service on my android system. >The application is Csipsimple. > >On Wifi and 3G connection, your service is very good. your sip account did >good work. even 3G network connection. >Btw, On Mobile Wimax service, the signal was good. But the Voice could not >be trasnsferred. > > >I guess the problem is about rtp port. >*I'd like to know the rtp port that IPTEL use*. I will request Korea Telecom >about rtp port open. >Korea Telecom is Korean mobile Wimax Provider. > >Thank you. > >Sungjin Park >Seoul, Korea >-------------- next part -------------- >An HTML attachment was scrubbed... >URL: <http://lists.sip-router.org/pipermail/sr-users/attachments/20110503/34b8728…> > >------------------------------ > >Message: 3 >Date: Tue, 3 May 2011 12:21:22 -0300 (ART) >From: Claudio Furrer <elcaio(a)gmail.com> >Subject: [SR-Users] INSTALL file >To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List" <sr-users(a)lists.sip-router.org> >Message-ID: <alpine.LNX.2.00.1105031219240.29861(a)askepios.local> >Content-Type: TEXT/PLAIN; charset=US-ASCII > >Hello, > >Is this still applicable? > >See INSTALL file (line 669) > >3.1) create MySQL tables for SER flavour > - if you have a previously installed SER on your system, use > /usr/sbin/ser_mysql.sh reinstall > to convert your SER database into new structures > - otherwise, if this is your very first installation, use > /usr/sbin/ser_mysql.sh create > to create SER database structures > (you will be prompted for password of MySql "root" user) > >Because I did not found the *reinstall* option on sources.. > >Thanks, >Claudio > > > >------------------------------ > >Message: 4 >Date: Tue, 3 May 2011 12:49:22 -0300 (ART) >From: Claudio Furrer <elcaio(a)gmail.com> >Subject: [SR-Users] INSTALL file >To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List" <sr-users(a)lists.sip-router.org> >Message-ID: <alpine.LNX.2.00.1105031223590.29861(a)askepios.local> >Content-Type: TEXT/PLAIN; charset=US-ASCII > >Hello, > >Is this still applicable? > >See INSTALL file (line 669) > >3.1) create MySQL tables for SER flavour > - if you have a previously installed SER on your system, use > /usr/sbin/ser_mysql.sh reinstall > to convert your SER database into new structures > - otherwise, if this is your very first installation, use > /usr/sbin/ser_mysql.sh create > to create SER database structures > (you will be prompted for password of MySql "root" user) > >Because I did not found the *reinstall* option on sources.. > >Thanks, >Claudio > > > >------------------------------ > >Message: 5 >Date: Wed, 4 May 2011 17:18:17 +0800 (CST) >From: hb <hhb.1988(a)163.com> >Subject: [SR-Users] RE?RE? ? about presence >To: bucur_marius_ovidiu(a)yahoo.com >Cc: sr-users(a)lists.sip-router.org >Message-ID: <694e6d8c.1c32b.12fba4cfaed.Coremail.hhb.1988(a)163.com> >Content-Type: text/plain; charset="gbk" > >Dear?Bucur Marius? > Thank you for your help. > As you said the presence module does not work if I disable it's module. This time I used wireshark to get the packages. I found when I disable the presence module, one point send the "Notify" packages directly to the other, and there were no changes in the watchers table. And when I enable the module, I found it can create some lists in watchers. But at this time , I found there is nothing contents in packages like "Subscribe" and "Notify".What's more when one point change the state, it dose not send any packages to the server. > You said active_watchers will have something when the subscribe is authorised. What dose it mean? My server have not authorised the subscribe? But it can have a call with other subscribe. > Look foreword your help. > haibo huang >-------------- next part -------------- >An HTML attachment was scrubbed... >URL: <http://lists.sip-router.org/pipermail/sr-users/attachments/20110504/3f6d4fb…> > >------------------------------ > >_______________________________________________ >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 72, Issue 5 >***************************************

1 0

RE：RE： ? about presence
by hb 04 May '11

04 May '11

Dear：Bucur Marius： Thank you for your help. As you said the presence module does not work if I disable it's module. This time I used wireshark to get the packages. I found when I disable the presence module, one point send the "Notify" packages directly to the other, and there were no changes in the watchers table. And when I enable the module, I found it can create some lists in watchers. But at this time , I found there is nothing contents in packages like "Subscribe" and "Notify".What's more when one point change the state, it dose not send any packages to the server. You said active_watchers will have something when the subscribe is authorised. What dose it mean? My server have not authorised the subscribe? But it can have a call with other subscribe. Look foreword your help. haibo huang

1 0

INSTALL file
by Claudio Furrer 03 May '11

03 May '11

Hello, Is this still applicable? See INSTALL file (line 669) 3.1) create MySQL tables for SER flavour - if you have a previously installed SER on your system, use /usr/sbin/ser_mysql.sh reinstall to convert your SER database into new structures - otherwise, if this is your very first installation, use /usr/sbin/ser_mysql.sh create to create SER database structures (you will be prompted for password of MySql "root" user) Because I did not found the *reinstall* option on sources.. Thanks, Claudio

1 0

： about presence
by hb 03 May '11

03 May '11

Dear everyone: I use kamalio 3.1. I want to add the xcap_sever and presence modules into my kamailio. I followed the lists "http://cache.baidu.com/c?m=9d78d513d99a12f00bb6ca255001d7175b14d7226b8b8142…" step by step. The only difference is that i use the database postgresql. When i start the server, it worked well, but the softphone was not able to show the presence of the contacts. I tried many times and found that when i disable the sentence "#!define WITH_PRECENCE" in "kamailio.cfg", it can show the state of the contacts. However, when i check the database, there was nothing in "Active Watchers" and "Presentity" tables. I think there must have something wrong with the configure of presence. Any response will be appreciated. ps: the configuration of the presence in kamailio.cfg as follows(same as theTutorial): #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) modparam("presence", "fallback2db", 1) modparam("presence", "db_update_period", 20) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 0) modparam("presence_xml", "integrated_xcap_server", 1) #!endif haibo huang

3 2

Error compiling Kamailio with mi_xmlrpc
by Lucas Alvarez 02 May '11

02 May '11

Hi, I'm trying to compile kamailio with mi_xmlrpc, I've followed the instructions, first installing xmlrpc-c-0.9.1.0 and xmlrpc-c-dev-0.9.1.0 but the folder xmlrpc-c in /usr/include is not created, so kamailio couldn't be compiled. Then I've installed the latest version of xmlrpc-c-dev, in this case the folder xmlrpc-c with the headers was created but the module didn;t compile, this are the errors: Makefile:46: You are using an unsupported libxmlrpc-c3 version (1.16.34), compile at your own risk! CC (gcc) [M mi_xmlrpc.so] abyss_data.o abyss_data.c:218: error: expected ')' before '*' token abyss_data.c:235: error: expected ')' before '*' token abyss_data.c:248: error: expected ')' before '*' token abyss_data.c:286: error: expected ')' before '*' token abyss_data.c:298: error: expected ')' before '*' token abyss_data.c:311: error: expected ')' before '*' token abyss_data.c:325: error: expected ')' before '*' token abyss_data.c:331: error: expected ')' before '*' token make: *** [abyss_data.o] Error 1 I would appreciate if someone can point me through the right path. Thanks in advance. Lucas Alvarez

1 0

i want to setup a SIP proxy and i dont know where to start.
by Eliezer Croitoru 02 May '11

02 May '11

i have a main office and a branch office. the main office and the branch office connected via vpn connection the branch office is natted to the main office. so i want to put on the gateway machine a SIP proxy server to nat the sip connection and also to proxy all the connections to SIP uses of the office. means i have one asterisk machine on the main site and other SIP providers that in a case i want to use i can use them also without connecting to the main office. i dont need asterisk on the branch office. the gateway server is ubuntu 10.04 amd64 what i should use Kamailio or the openser on the ubuntu repos? for now the Kamailio is installed.(not working). another feature i want to add is to be able to make calls between SIP phones using this server such as berni(a)mydomain.com from outside the office and vise verse... to dial people at bob(a)otherdomain.com hope for little help Eliezer

2 2

Kamailio performance testing
by Martin Pohančeník 02 May '11

02 May '11

Hi I am trying to test performance of kamailio server. I have made some tests on old hardware, that was ok. I found that it can handle around 450 calls per second with authorization. CPU utilisation was up to 100%. That testing was OK, because server's CPU utilization was increasing. Now I want to do some test with new hardware and do some comparisons of those two hardware configurations. I have made some test but i found that it can handle up to 540 calls per second with authorization. But CPU utilisation was only around 8%. When I increased call rate server responded with 404 Not Here and 500 I am terribly busy messages. I think that it is because of that it is not capable of finding a user location that fast. Can i do something to increase the performacne of this server? I am using SIPp testing tool for generating calls. I have registered users manualy with kamctl ul add sipp sip:sipp@192.168.1.1:5060 command. I was using this tutorial: http://www.troubleshootingwiki.org/OpenSER#Stress_Test_The_SIP_Signaling. And also i made my own scenarios. Any suggestion would be usefull. Thanks.

1 0

New module ipops
by Iñaki Baz Castillo 02 May '11

02 May '11

Hi, I've commited a new module named 'ipops' which provides operations with IPv4 and IPv6: http://git.sip-router.org/cgi-bin/gitweb.cgi/sip-router/?a=commit;h=5940394… The README describes the module functions: http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=blob;f=modules/… The module has no dependecies on other siprouter modules or external libraries, so if it's ok for you it could be enabled by default. Cheers. -- Iñaki Baz Castillo <ibc(a)aliax.net>

3 3

SIP outbound
by Klaus Darilion 02 May '11

02 May '11

Hi! IIRC we recently had a discussion about SIP outbound [1] and the benefits of an implementation in Kamailio. Just playing around with a SIP client on an iphone I discovered a usecase. Whenever the iphone gets a new IP address (quite often as it often changes between WiFi and 3G) the client reREGISTERs with the new IP address (establishing a new TCP connection). This leads to lots of "broken" registrations and TCP connections [2]. By use of SIP outbound and re-registration using the same "reg-id" the proxy would be able to replace the old registration with the new registration. Further, if Kamailio would have a mapping from registrations to TCP connections it could also close the dead TCP connections. regards Klaus [1] http://tools.ietf.org/html/rfc5626 [2] I do not know if TCP keepalives (socket option) could detect broken TCP connections without making too much noise on the receiver side (activating the background application and draining battery)

4 4

Kamailio v3.1.3 in Action
by Andreas Granig 01 May '11

01 May '11

Hi all, If you want to see a full-blown Kamailio v3.1.3 based open-source VoIP system in action, check our latest sip:provider CE v2.2-rc1, which has been released today: http://www.sipwise.com/news/announcements/spce-v2_2rc1-release/ It leverages the capabilities of Kamailio to build a quite complex SIP routing engine using a SIP load-balancer and a SIP proxy/registrar, as well as SEMS acting as b2bua. On top of it, it provides provisioning and billing modules and admin and user web interfaces to form an out-of-the-box VoIP soft-switch. Feedback is very much appreciated. Andreas

1 0

uri_db table version
by Claudio Furrer 29 Apr '11

29 Apr '11

Hello, I've found that scripts/my_create.sql doesn't create the version table anymore (new 3.1 installation). It may be removed since many versions ago, sorry i don't know. Loading the modules/db_mysql and modules_k/uri_db, with a SER flavour compilation (and also a ser config file), it raises: 0(15262) DEBUG: db_mysql [km_my_con.c:94]: opening connection: mysql://xxxx:xxxx@localhost/srtest 0(15262) DEBUG: db_mysql [km_my_con.c:121]: connection type is Localhost via UNIX socket 0(15262) DEBUG: db_mysql [km_my_con.c:122]: protocol version is 10 0(15262) DEBUG: db_mysql [km_my_con.c:123]: server version is 5.1.51-log 0(15262) ERROR: db_mysql [km_dbase.c:120]: driver error on query: Table 'ser_new.version' doesn't exist 0(15262) ERROR: <core> [db_query.c:101]: error while submitting query 0(15262) ERROR: <core> [db.c:366]: error in db_query Should I create the version table manually or using an old script? Thank you. Regards, Claudio

3 5

Re: [SR-Users] How to get rid of this error message.
by Gary Chen 29 Apr '11

29 Apr '11

Thanks. This fixed error. Is $rc the same as $retcode ? Gary -----Original Message----- From: Daniel-Constantin Mierla [mailto:miconda@gmail.com] Sent: Friday, April 29, 2011 1:01 PM To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List Cc: Gary Chen Subject: Re: [SR-Users] How to get rid of this error message. Hello, On 4/29/11 4:32 PM, Gary Chen wrote: > I am using prefix2domain("2") in PDT module like this: > $var(found) = prefix2domain("2"); > and I got this error: > ERROR: <core> [lvalue.c:411]: assignment failed at pos: > (701,56-701,56) Apparently when prefix2domain("2) returns an value to > $var(found), it does not like it. But it still work. > The $var(found) still contains the return value (-1) if no prefix > found. And all my configuration is working. It just generate a lot of > this kind of error in my log file. Am I doing something wrong? How can > I fix it? I will look at it, it might be that the -1 return code is usually associated with an error case. Meanwhile try to replace that line with the following: prefix2domain("2"); $var(found) = $rc; Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

2 1

How to get rid of this error message.
by Gary Chen 29 Apr '11

29 Apr '11

I am using prefix2domain("2") in PDT module like this: $var(found) = prefix2domain("2"); and I got this error: ERROR: <core> [lvalue.c:411]: assignment failed at pos: (701,56-701,56) Apparently when prefix2domain("2) returns an value to $var(found), it does not like it. But it still work. The $var(found) still contains the return value (-1) if no prefix found. And all my configuration is working. It just generate a lot of this kind of error in my log file. Am I doing something wrong? How can I fix it? Gary

3 2

Kamailio with RTPoxy
by alex pappas 29 Apr '11

29 Apr '11

Dear all, I have a Kamailio with RTPoxy configured. The purpose is to pass the RTP from this server. When I make calls behind NAT the RTProxy seems to be engaged but in traces I don't see any RTP. Is there anything wrong with the idea or I'm missing something in the configuration ? Thank you Alex

3 5

is $td.did avp a normal avp?
by Juha Heinanen 26 Apr '11

26 Apr '11

domain module sets $td.did at lookup_domain call. $td.did seems to be a normal avp, because i can print its value with xlog by referring to it as $avp(td.did). however, its value is gone at failure route, which makes me suspect that $td.did is not really a normal avp. in route block i have $avp(did) = $td.did; xlog("L_INFO", "at t_relay: <td/td.did> = <$avp(did)/$avp(td.did)>\n"); if (t_relay()) exit; branch route, onreply route, and failure route are set when the above t_relay() takes place. at the end of branch route block, i have xlog("L_INFO", "at end of contact branches: <did/td.did> = <$avp(did)/$avp(td.did)>\n"); and at the beginning of onreply route block: xlog("L_INFO", "at beginning of onreply: <did/td.did> = <$avp(did)/$avp(td.did)>\n"); and at the beginning of failure route block: xlog("L_INFO", "at beginning of contact failure: <did/td.did = <$avp(did)/$avp(td.did)>\n"); when i make a call, i get to syslog: Apr 22 15:53:27 sip /usr/sbin/sip-proxy[13706]: INFO: at t_relay: <td/td.did> = <test.fi/test.fi> Apr 22 15:53:27 sip /usr/sbin/sip-proxy[13706]: INFO: at end of contact branches: <did/td.did> = <test.fi/test.fi> Apr 22 15:53:27 sip /usr/sbin/sip-proxy[13708]: INFO: at beginning of onreply: <did/td.did> = <test.fi/<null>> Apr 22 15:53:27 sip /usr/sbin/sip-proxy[13705]: INFO: at beginning of contact failure: <did/td.did = <test.fi/<null>> why is it that $avp(did) value has been preserved from route/branch route block to onreply/failure blocks, but value of $avp(td.did) is not? i'll cc to jan in case he is not on this list. -- juha

4 12

call for participation: kamailio project at LinuxTag 2011
by Henning Westerholt 26 Apr '11

26 Apr '11

Hello, in about five weeks the kamailio project will (again) be present on the LinuxTag fair, "the most important place for Linux and open source software in Europe" - http://www.linuxtag.org/2011/en.html . If you are in Berlin during this time, plan to visit the LinuxTag or just like to help your favourite SIP server project - please consider supporting us in running our fair booth! Even showing up for one or two hours helps a lot. We need also support to properly present our project, like flyers, a banner, maybe some live demonstrations.. So even if you can't participate in person, there are other ways to help us! Please contact me per mail if you could help at the booth or have some ideas about our project presentation. The actual state of preparations can be found in our wiki: http://sip-router.org/wiki/meetings/berlin_2011 Cheers, Henning

1 1

Bug in LDAP module?
by Pan B. Christensen 26 Apr '11

26 Apr '11

Hello, We are using kamailio 1.5 and we have the following problem with ldap_search: A call comes in. If it is to a VOIP user, it will be routed to kamailio, and eventually go to the ldap lookup route: route[10] to look up the user. This is always successful (LDAP can connect). It will then look up user preferences (call forwarding etc.) from SQL in route[12] and route based on this result in route[13]. If the result is "CFU (call forward unconditional) = on", the URI is changed and route[10] (LDAP lookup) is called again. This is always successful (LDAP can connect). If the result is "CFB (call forward busy) = on" and the user is busy, the URI is changed and route[10] (LDAP lookup) is called again. This is also always successful (LDAP can connect). If the result is "CFNR (call forward no reply) = on", the timeout is changed and the call is delivered to the phone through route[14] (location) and route[15] (relay). If the user does not answer, the URI is changed in failure_route[1] and route[10] is called. For some unknown reason, ldap_search always fails in this scenario with the messages: "ldap_session not found", "couldn't get ldap session" and then "no LDAP entry found". Why? And why does this result in the -1 return code and not the -2 return code for ldap_search? The only differences I can think of is that the unsuccessful scenario comes from FAILURE_ROUTE instead of REQUEST_ROUTE, and that longer time has passed since the original invite. Here's a sample log: NOTE the three ERROR lines, the INFO line and the ROUTE 10 line. Dec 27 13:31:55 [3373]: INVITE received from sip:<CALLER>@<SBC_IP> <source IP: <SBC_IP>> to sip:<CALLEE>@<KAM_IP>: Dec 27 13:31:55 [3373]: ERROR:ldap:ldap_reconnect: [ldapprod]: LDAP reconnect successful Dec 27 13:31:55 [3373]: ROUTE 10: LDAP returned: UID = <UID>, voipSipPrimaryUri = <PRIMARY_URI>, voipExtensionUri = sip:<CALLEE>@<DOMAIN> Dec 27 13:31:55 [3373]: ROUTE 12: Effective user properties for UID <UID>: CFU: off <FORWARDTO>, CFB: off <FORWARDTO>, CFNR: on:25 <FORWARDTO>, CFNC: off <FORWARDTO>, VM: <CALLEE>, busy_level: 1, DBrows: 1 Dec 27 13:31:55 [3373]: -------------------- Calling sip:<CALLEE>@<DOMAIN> for 25 seconds. -------------------- Dec 27 13:31:55 [3373]: ROUTE 15: Sending 105 INVITE from sip:<CALLER>@<SBC_IP> to sip:<CALLEE>@<CALLEE_IP> (via <<null>>) Dec 27 13:31:55 [3383]: Response 100 Trying to Cseq 102 INVITE received from <PROTO>:<CALLEE_IP>:<PORT> Dec 27 13:31:55 [3385]: Response 180 Ringing to Cseq 102 INVITE received from <PROTO>:<CALLEE_IP>:<PORT> Dec 27 13:32:20 [3388]: FAILURE_ROUTE 1: Timeout for: 104 INVITE from <CALLER> -> <CALLEE> Dec 27 13:32:20 [3388]: -------------------- Forwarding on no reply to <FORWARDTO> -------------------- Dec 27 13:32:20 [3388]: ERROR:ldap:get_connected_ldap_session: [ldapprod]: ldap_session not found Dec 27 13:32:20 [3388]: ERROR:ldap:lds_search: [ldapprod]: couldn't get ldap session Dec 27 13:32:20 [3388]: INFO:ldap:ldap_search_impl: no LDAP entry found Dec 27 13:32:20 [3388]: ROUTE 11: Number lookup results for <FORWARDTO>: Node: VOIP. Dec 27 13:32:20 [3388]: CONFIG ERROR: VOIP number not found in LDAP. Hanging up. Dec 27 13:32:20 [3373]: Response 487 Request Cancelled to Cseq 102 INVITE received from <PROTO>:<CALLEE_IP>:<PORT> Here's the ldap config file: [ldapprod] ldap_server_url = "ldap://ldap.<DOMAIN>" ldap_network_timeout = 500 ldap_client_bind_timeout = 500 kamailio config snippets (in case you need them): # LDAP lookup for UID and primary URI route[10] { ... ldap_search("ldap://ldapprod/<DN>?uid,voipSipPrimaryUri,voipExtensionUri?one?(&(objectClass=voipAddress)(|(voipSipUri=sip:$rU@uio.no)(|(voipSipUri=$var(uri))(voipSipUri=$var(e164uri)))))"); if ($retcode < 1) { switch ($retcode) { case -1: # no LDAP entry found route(11); return(1); case -2: # internal error sl_send_reply("500", "Internal server error"); exit; } } ldap_result("uid/$avp(s:uid)"); $avp(s:exten) = null; if (!ldap_result("voipExtensionUri/$avp(s:exten)")) { xlog("L_INFO", "voipExtensionUri not found.\n"); } if (!ldap_result("voipSipPrimaryUri/$avp(s:p-uri)")) { xlog("L_WARN", "voipSipPrimaryUri not found.\n"); #exit here? } if (isflagset(1)) { xlog("L_INFO", "ROUTE 10: LDAP returned: UID = $avp(s:uid), voipSipPrimaryUri = $avp(s:p-uri), voipExtensionUri = $avp(s:exten)\n"); } if $avp(s:exten) != null { $ru=$avp(s:exten); } else { $ru=$avp(s:p-uri); } route(12); } } # Number lookup. Non-VoIP URI's go here. route[11] { ... } # Database lookup for call forwarding. route[12] { ... } # Call forwarding logic route[13] { if (is_method("INVITE")) { $avp(s:uio_timer_avp) = "300"; if ($avp(s:cfu_status) == "on") { # Forward immediately xlog("L_INFO", "-------------------- $rU forwarded unconditionally to $avp(s:cfu_number). --------------------\n"); $rU = $avp(s:cfu_number); route(10); } ... if ( $var(dlg_busy) >= $avp(s:busy_level) ) { if ($avp(s:cfb_status) == "on") { xlog("L_INFO", "-------------------- $rU has $var(dlg_busy) active calls. Treshold $avp(s:busy_level). Forwarding on busy to $avp(s:cfb_number) --------------------\n"); $rU = $avp(s:cfb_number); route(10); } ... else if ($avp(s:cfnr_status) == "on") { # Forward to phone with timeout then to $avp(s:cfnr_number). # t_set_fr($avp(s:cfnr_timeout) + "000"); # Recommended method in 3.1. Function does not exist in 1.5. Using old method (avp). $avp(s:uio_timer_avp) = $avp(s:cfnr_timeout); xlog("L_INFO", "-------------------- Calling $ru for $avp(s:uio_timer_avp) seconds. --------------------\n"); route(14); } ... } failure_route[1] { ... if $avp(s:cfnr_status) == "on" { if (isflagset(1)) { xlog("L_INFO", "FAILURE_ROUTE 1: Timeout for: $mi $rm from $fU -> $rU\n"); } revert_uri(); $rU = $avp(s:cfnr_number); xlog("L_INFO", "-------------------- Forwarding on no reply to $rU --------------------\n"); route(10); } ... } With kind regards, Pan

3 5

Kamailio stops responding after 10 days or so
by Morten Isaksen 26 Apr '11

26 Apr '11

Hi! Kamailio 3.0.3. I have a strange problem with one of our Kamailio servers. This one is used for routing (with carrierroute) and to send presence information (with pua module) Once every 10 day or so I get this error and then Kamailio stops responding to any SIP packets. Apr 6 08:05:48 sip-core-1 /usr/local/sbin/kamailio[9186]: WARNING: <script>: Failure route - M=INVITE RURI=sip:8615XXXX@178.XX.XX.XX F=sip:861XXXXX@188.120.93.114:1025 T=sip:86155XXXX@sip1.uni-tel.dk IP=178.XX.XX.XX ID=6de881ec07f9c6494ee589cf208da358(a)10.11.87.206 Apr 6 08:05:48 sip-core-1 /usr/local/sbin/kamailio[9186]: ERROR: carrierroute [cr_func.c:95]: cannot find AVP 'carrier' Apr 6 08:05:48 sip-core-1 /usr/local/sbin/kamailio[9186]: ERROR: carrierroute [cr_func.c:805]: invalid carrier id -1 Apr 6 08:05:48 sip-core-1 /usr/local/sbin/kamailio[9186]: ERROR: <script>: cr_next_domain failed Shared memory size is 128M and over halv is free just before the error. The server is in production and does handle debug>1 well, so I do not have much information in the log files. Private memory is the default size. Any ideas what it could be, or how to investegate further? I think my next steps would be to increase the private memory og to increase children=4 to children=8 -- Morten Isaksen

3 7

Command 'sercmd lcr.dump_rules' out of memory.
by Alexandre Abreu 26 Apr '11

26 Apr '11

Hello. When running 'sercmd lcr.reload' (without any errors) and then 'sercmd lcr.dump_rules' I've got the following message: ERROR: ctl [binrpc_run.c:510]: ERROR: binrprc: rpc_send: too many message chunks Then I increased the CTL module params 'binrpc_max_body_size' and ' binrpc_struct_max_body_size'. Re-run the command: ERROR: ctl [binrpc_run.c:908]: ERROR: binrpc: rpc_add: out of memory kamailio -V version: kamailio 3.1.3 (x86_64/linux) flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: compiled on 19:43:43 Apr 18 2011 with gcc 4.1.2 It doesn't matter if I increase the PKG_SIZE (> 8). The LCR tables are very small. lcr_rule - 445 lines lcr_rule_target - 445 lines lcr_gw - 7 lines Alexandre

2 1

Re: [SR-Users] Ipv6 SIP UA
by Klaus Darilion 26 Apr '11

26 Apr '11

You are right! There are actually 2 issues with pjsip: - UDP6 transport does not support DNS lookups. Thus you must set an outboundproxy with IPv6 address - pjsua API does not handle IPv6 media (SDP) As Qjsimple is based on pjsip's pjsua API QjSimple of course has the same issue. regards klaus Am 21.04.2011 14:05, schrieb Pavel Segeč: > Klaus, > > still the same, > > here is a log when UDP is used, there is only A record resolving > > QjSimple: starting initialization/registration ... > QjSimple: initializing SIP stack ... > QjSimple: pjsua_create succeeded ... > QjSimple: "qjsimple" memory pool created ... > QjSimple: DLL loaded ... > QjSimple: GetNetworkParams found ... > QjSimple: GetNetworkParams done ... > QjSimple: cfg.nameserver_count=0 > QjSimple: pjsua_init succeeded ... > QjSimple: pjsip_endpt_register_module succeeded ... > QjSimple: adding UDP transport ... > QjSimple: pjsip_endpt_register_module succeeded ... > QjSimple: outboundUri=sip:ps.sip.uniza.sk;lr;transport=udp > QjSimple: registration_enabled=TRUE > QjSimple: publish_enabled=FALSE > QjSimple: pjsua_acc_add succeeded ... > QjSimple: finished intializing SIP stack, start pjsua ... > QjSimple: Info: on_reg_state_slot called > QjSimple: Info: on_reg_state_slot: status=502 > QjSimple: Info: on_reg_state_slot: statustext=Bad Gateway > QjSimple: Info: on_reg_state_slot called > QjSimple: Info: on_reg_state_slot: status=502 > QjSimple: Info: on_reg_state_slot: statustext=Bad Gateway > > and here when UDP6 > > QjSimple: starting initialization/registration ... > QjSimple: initializing SIP stack ... > QjSimple: pjsua_create succeeded ... > QjSimple: "qjsimple" memory pool created ... > QjSimple: DLL loaded ... > QjSimple: GetNetworkParams found ... > QjSimple: GetNetworkParams done ... > QjSimple: cfg.nameserver_count=0 > QjSimple: pjsua_init succeeded ... > QjSimple: pjsip_endpt_register_module succeeded ... > QjSimple: adding UDP6 transport ... > QjSimple: pjsip_endpt_register_module succeeded ... > QjSimple: outboundUri=sip:ps.sip.uniza.sk;lr;transport=udp > QjSimple: registration_enabled=TRUE > QjSimple: publish_enabled=FALSE > QjSimple: pjsua_acc_add succeeded ... > QjSimple: finished intializing SIP stack, start pjsua ... > > there is no DNS at all > > pavel > > >> -----Original Message----- >> From: sr-users-bounces(a)lists.sip-router.org [mailto:sr-users- >> bounces(a)lists.sip-router.org] On Behalf Of Klaus Darilion >> Sent: Thursday, April 21, 2011 1:27 PM >> To: sr-users(a)lists.sip-router.org >> Subject: Re: [SR-Users] Ipv6 SIP UA >> >> Pavel, I have updated QjSimple to latest pjsip version (0.8.10). It >> would be great if you can test again. If the problem still exists I will >> post a bug report for pjsip. >> >> http://www.ipcom.at/fileadmin/ipcom/files/qjsimple/QjSimple-0.6.6.zip >> >> regards >> Klaus >> >> Am 21.04.2011 10:10, schrieb Pavel Segeč: >>> Hi Klaus, >>> >>> I'm testing your QJSimple client and I'm observing following problems. >> In a IPv4 environments (win 7/32 or 64bit), the client is using a >> classical DNS resolving with SRV and so on, and I'm able to register. >> But inside of the IPV6 environment (IPv4 disabled), when UDP transport >> is used, the client is making just the A DNS record resolving on my >> domain name, not SRV at all. >>> >>> Here is debug log: >>> >>> QjSimple: starting registration ... >>> QjSimple: initializing SIP stack ... >>> QjSimple: pjsua_create succeeded ... >>> QjSimple: "qjsimple" memory pool created ... >>> QjSimple: DLL loaded ... >>> QjSimple: GetNetworkParams found ... >>> QjSimple: GetNetworkParams done ... >>> QjSimple: cfg.nameserver_count=0 >>> QjSimple: pjsua_init succeeded ... >>> QjSimple: pjsip_endpt_register_module succeeded ... >>> QjSimple: adding UDP transport ... >>> QjSimple: pjsip_endpt_register_module succeeded ... >>> QjSimple: outboundUri=sip:ps.sip.uniza.sk;lr;transport=udp >>> QjSimple: publish_enabled=FALSE >>> QjSimple: pjsua_acc_add succeeded ... >>> QjSimple: finished intializing SIP stack, start pjsua ... >>> QjSimple: initializing SIP stack ... done >>> QjSimple: Info: on_reg_state_slot called >>> QjSimple: Info: on_reg_state_slot: status=502 >>> QjSimple: Info: on_reg_state_slot: statustext=Bad Gateway >>> >>> When I setup as the transport protocol UDP6 there is no actions at >> all, no dns, no sip, he re is debug log >>> >>> QjSimple: starting registration ... >>> QjSimple: initializing SIP stack ... >>> QjSimple: pjsua_create succeeded ... >>> QjSimple: "qjsimple" memory pool created ... >>> QjSimple: DLL loaded ... >>> QjSimple: GetNetworkParams found ... >>> QjSimple: GetNetworkParams done ... >>> QjSimple: cfg.nameserver_count=0 >>> QjSimple: pjsua_init succeeded ... >>> QjSimple: pjsip_endpt_register_module succeeded ... >>> QjSimple: adding UDP6 transport ... >>> QjSimple: pjsip_endpt_register_module succeeded ... >>> QjSimple: outboundUri=sip:ps.sip.uniza.sk;lr;transport=udp >>> QjSimple: publish_enabled=FALSE >>> QjSimple: pjsua_acc_add succeeded ... >>> QjSimple: finished intializing SIP stack, start pjsua ... >>> QjSimple: initializing SIP stack ... done >>> >>> regards >>> >>> pavel >>> >>>> -----Original Message----- >>>> From: sr-users-bounces(a)lists.sip-router.org [mailto:sr-users- >>>> bounces(a)lists.sip-router.org] On Behalf Of Klaus Darilion >>>> Sent: Wednesday, April 20, 2011 12:49 PM >>>> To: sr-users(a)lists.sip-router.org >>>> Subject: Re: [SR-Users] Ipv6 SIP UA >>>> >>>> >>>> >>>> Am 20.04.2011 11:01, schrieb Pavel Segeč: >>>>> Hi, >>>>> >>>>> yes I can agree now, linphone is working well, qjsimple has some >>>> problem >>>> >>>> If you find some problems (except that it does not work over TCL/TLS: >>>> http://trac.pjsip.org/repos/ticket/1127) please let me know. >>>> >>>> klaus >>>> >>>> _______________________________________________ >>>> 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 >>>> >>>> >>>> __________ Informacia od ESET NOD32 Antivirus, verzia databazy 6057 >>>> (20110420) __________ >>>> >>>> Tuto spravu preveril ESET NOD32 Antivirus. >>>> >>>> http://www.eset.sk >>>> >>>> >>>> __________ Informacia od ESET NOD32 Antivirus, verzia databazy 6059 >>>> (20110420) __________ >>>> >>>> Tuto spravu preveril ESET NOD32 Antivirus. >>>> >>>> http://www.eset.sk >>>> >>> >>> >>> __________ Informacia od ESET NOD32 Antivirus, verzia databazy 6059 >> (20110420) __________ >>> >>> Tuto spravu preveril ESET NOD32 Antivirus. >>> >>> http://www.eset.sk >>> >>> >> >> _______________________________________________ >> 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 >> >> >> __________ Informacia od ESET NOD32 Antivirus, verzia databazy 6060 >> (20110421) __________ >> >> Tuto spravu preveril ESET NOD32 Antivirus. >> >> http://www.eset.sk >> > > > __________ Informacia od ESET NOD32 Antivirus, verzia databazy 6060 (20110421) __________ > > Tuto spravu preveril ESET NOD32 Antivirus. > > http://www.eset.sk > > > > _______________________________________________ > 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

3 2

Try backup route
by qmail＠top-consulting.net 22 Apr '11

22 Apr '11

I'm using Kamailio 3.1 I have a custom routing statement called MYROUTE which is placed like this: route { ... route (MYROUTE); route (RELAY);} Inside MYROUTE I do some custom SQL checking that basically selects the least cost provider for the ongoing call. The MYROUTE routine sets the $rU, $rd and $rp variables in case a provider has been found for the dialed number or fails with Sip code 404. All this works perfectly, except in my database I also have an alternate provider that I only wish to use as backup in case the main provider returns a 5XX code. How do I get Kamailio to try the second provider after the first one failed ? So far I've tried setting up a t_on_failure() command & routine that picks up the backup provider from the database and sets $rU, $rd margin-right:0px;margin-bottom:0px;margin-left:0px;padding-top:0px;padding-right:0px;padding-bottom:0px;padding-left:0px;"> 0(11460) DEBUG: [parser/msg_parser.c:167]: get_hdr_field: cseq : 0(11460) DEBUG: [parser/msg_parser.c:201]: DEBUG: get_hdr_body : content_length=0 0(11460) DEBUG: [parser/msg_parser.c:103]: found end of header 0(11460) DEBUG: maxfwd [mf_funcs.c:66]: max_forwards header not found! 0(11460) DEBUG: [parser/parse_to.c:174]: DEBUG: add_param: tag=4a2aab60 0(11460) DEBUG: [parser/parse_to.c:803]: end of header reached, state=29 0(11460) DEBUG: siputils [checks.c:76]: totag found 0(11460) DEBUG: rr [loose.c:108]: No Route headers found 0(11460) DEBUG: rr [loose.c:910]: There is no Route HF I believe the problem is that t_on_failure() does not keep all the proper data associated with the call and that's why it fails. Maybe t_on_failure isn't the right command to use here after all (?). So to resume. Here's what I wish to achieve: 1 Call comes through2. Goes through MYROUTE where a provider is selected3. Call is attempted through provider found in #24. If call fails , select another provider5. Call is attempted through provider found in #4 Thanks!

2 1

Try backup route
by qmail＠polarismail.com 21 Apr '11

21 Apr '11

I'm using Kamailio 3.1 I have a custom routing statement called MYROUTE which is placed like this: route { ... route (MYROUTE); route (RELAY);} Inside MYROUTE I do some custom SQL checking that basically selects the least cost provider for the ongoing call. The MYROUTE routine sets the $rU, $rd and $rp variables in case a provider has been found for the dialed number or fails with Sip code 404. All this works perfectly, except in my database I also have an alternate provider that I only wish to use as backup in case the main provider returns a 5XX code. How do I get Kamailio to try the second provider after the first one failed ? So far I've tried setting up a t_on_failure() command & routine that picks up the backup provider from the database and sets $rU, $rd margin-right:0px;margin-bottom:0px;margin-left:0px;padding-top:0px;padding-right:0px;padding-bottom:0px;padding-left:0px;"> 0(11460) DEBUG: [parser/msg_parser.c:167]: get_hdr_field: cseq : 0(11460) DEBUG: [parser/msg_parser.c:201]: DEBUG: get_hdr_body : content_length=0 0(11460) DEBUG: [parser/msg_parser.c:103]: found end of header 0(11460) DEBUG: maxfwd [mf_funcs.c:66]: max_forwards header not found! 0(11460) DEBUG: [parser/parse_to.c:174]: DEBUG: add_param: tag=4a2aab60 0(11460) DEBUG: [parser/parse_to.c:803]: end of header reached, state=29 0(11460) DEBUG: siputils [checks.c:76]: totag found 0(11460) DEBUG: rr [loose.c:108]: No Route headers found 0(11460) DEBUG: rr [loose.c:910]: There is no Route HF I believe the problem is that t_on_failure() does not keep all the proper data associated with the call and that's why it fails. Maybe t_on_failure isn't the right command to use here after all (?). So to resume. Here's what I wish to achieve: 1 Call comes through2. Goes through MYROUTE where a provider is selected3. Call is attempted through provider found in #24. If call fails , select another provider5. Call is attempted through provider found in #4 Thanks!

1 0

BLF using kamailio
by Lucas Alvarez 21 Apr '11

21 Apr '11

Hi, I have implemented kamailio with asterisk, I want to enable BLF using kamailio instead of asterisk. I would appreciate any kind of help, below I'm pasting my configuration. *I'm testing with an Aastra phone, when phone sends the subscription kamailio answers with:* U 192.168.15.22:5060 -> 192.168.15.108:5060 SIP/2.0 489 Bad Event. Via: SIP/2.0/UDP 192.168.15.108;branch=z9hG4bKa54a155ad67706dca. From: "1104" <sip:1104@192.168.15.22:5060>;tag=71cf5c9949. To: <sip:1104@192.168.15.22:5060>;tag=99b712eabe88b2bf7a20409f3dc7ebf1-f71b. Call-ID: f22823b8b4d58a8e. CSeq: 3430 SUBSCRIBE. Allow-Events: presence.winfo, presence, dialog. Server: kamailio (3.1.3 (i386/linux)). Content-Length: 0. *This is my presence configuration:* #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_dialoginfo.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) modparam("presence", "server_address", "sip:192.168.15.22:5060") modparam("presence_dialoginfo", "force_single_dialog", 1) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif

4 5

New developer: Jason Penton
by Daniel-Constantin Mierla 21 Apr '11

21 Apr '11

Hello, I want to announce Jason Penton as a new registered developer, with write access to GIT repository - his username id is 'jason.penton'. Jason already submitted an IMS-related module that Carsten added to his IMS branch in the last days, therefore expect contributions from Jason in the IMS extensions. Thanks, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

2 1

Re: [SR-Users] Ipv6 SIP UA
by Klaus Darilion 21 Apr '11

21 Apr '11

Pavel, I have updated QjSimple to latest pjsip version (0.8.10). It would be great if you can test again. If the problem still exists I will post a bug report for pjsip. http://www.ipcom.at/fileadmin/ipcom/files/qjsimple/QjSimple-0.6.6.zip regards Klaus Am 21.04.2011 10:10, schrieb Pavel Segeč: > Hi Klaus, > > I'm testing your QJSimple client and I'm observing following problems. In a IPv4 environments (win 7/32 or 64bit), the client is using a classical DNS resolving with SRV and so on, and I'm able to register. But inside of the IPV6 environment (IPv4 disabled), when UDP transport is used, the client is making just the A DNS record resolving on my domain name, not SRV at all. > > Here is debug log: > > QjSimple: starting registration ... > QjSimple: initializing SIP stack ... > QjSimple: pjsua_create succeeded ... > QjSimple: "qjsimple" memory pool created ... > QjSimple: DLL loaded ... > QjSimple: GetNetworkParams found ... > QjSimple: GetNetworkParams done ... > QjSimple: cfg.nameserver_count=0 > QjSimple: pjsua_init succeeded ... > QjSimple: pjsip_endpt_register_module succeeded ... > QjSimple: adding UDP transport ... > QjSimple: pjsip_endpt_register_module succeeded ... > QjSimple: outboundUri=sip:ps.sip.uniza.sk;lr;transport=udp > QjSimple: publish_enabled=FALSE > QjSimple: pjsua_acc_add succeeded ... > QjSimple: finished intializing SIP stack, start pjsua ... > QjSimple: initializing SIP stack ... done > QjSimple: Info: on_reg_state_slot called > QjSimple: Info: on_reg_state_slot: status=502 > QjSimple: Info: on_reg_state_slot: statustext=Bad Gateway > > When I setup as the transport protocol UDP6 there is no actions at all, no dns, no sip, he re is debug log > > QjSimple: starting registration ... > QjSimple: initializing SIP stack ... > QjSimple: pjsua_create succeeded ... > QjSimple: "qjsimple" memory pool created ... > QjSimple: DLL loaded ... > QjSimple: GetNetworkParams found ... > QjSimple: GetNetworkParams done ... > QjSimple: cfg.nameserver_count=0 > QjSimple: pjsua_init succeeded ... > QjSimple: pjsip_endpt_register_module succeeded ... > QjSimple: adding UDP6 transport ... > QjSimple: pjsip_endpt_register_module succeeded ... > QjSimple: outboundUri=sip:ps.sip.uniza.sk;lr;transport=udp > QjSimple: publish_enabled=FALSE > QjSimple: pjsua_acc_add succeeded ... > QjSimple: finished intializing SIP stack, start pjsua ... > QjSimple: initializing SIP stack ... done > > regards > > pavel > >> -----Original Message----- >> From: sr-users-bounces(a)lists.sip-router.org [mailto:sr-users- >> bounces(a)lists.sip-router.org] On Behalf Of Klaus Darilion >> Sent: Wednesday, April 20, 2011 12:49 PM >> To: sr-users(a)lists.sip-router.org >> Subject: Re: [SR-Users] Ipv6 SIP UA >> >> >> >> Am 20.04.2011 11:01, schrieb Pavel Segeč: >>> Hi, >>> >>> yes I can agree now, linphone is working well, qjsimple has some >> problem >> >> If you find some problems (except that it does not work over TCL/TLS: >> http://trac.pjsip.org/repos/ticket/1127) please let me know. >> >> klaus >> >> _______________________________________________ >> 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 >> >> >> __________ Informacia od ESET NOD32 Antivirus, verzia databazy 6057 >> (20110420) __________ >> >> Tuto spravu preveril ESET NOD32 Antivirus. >> >> http://www.eset.sk >> >> >> __________ Informacia od ESET NOD32 Antivirus, verzia databazy 6059 >> (20110420) __________ >> >> Tuto spravu preveril ESET NOD32 Antivirus. >> >> http://www.eset.sk >> > > > __________ Informacia od ESET NOD32 Antivirus, verzia databazy 6059 (20110420) __________ > > Tuto spravu preveril ESET NOD32 Antivirus. > > http://www.eset.sk > >

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Re: [SR-Users] Ipv6 SIP UA
by Pavel Segeč 21 Apr '11

21 Apr '11

Hi, and the IMS Boghe client can be used for IPv6 call testing too. pavel > -----Original Message----- > From: sr-users-bounces(a)lists.sip-router.org [mailto:sr-users- > bounces(a)lists.sip-router.org] On Behalf Of Pavel Segeč > Sent: Thursday, April 21, 2011 10:10 AM > To: 'SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - > Users Mailing List' > Subject: Re: [SR-Users] Ipv6 SIP UA > > Hi Klaus, > > I'm testing your QJSimple client and I'm observing following problems. > In a IPv4 environments (win 7/32 or 64bit), the client is using a > classical DNS resolving with SRV and so on, and I'm able to register. > But inside of the IPV6 environment (IPv4 disabled), when UDP transport > is used, the client is making just the A DNS record resolving on my > domain name, not SRV at all. > > Here is debug log: > > QjSimple: starting registration ... > QjSimple: initializing SIP stack ... > QjSimple: pjsua_create succeeded ... > QjSimple: "qjsimple" memory pool created ... > QjSimple: DLL loaded ... > QjSimple: GetNetworkParams found ... > QjSimple: GetNetworkParams done ... > QjSimple: cfg.nameserver_count=0 > QjSimple: pjsua_init succeeded ... > QjSimple: pjsip_endpt_register_module succeeded ... > QjSimple: adding UDP transport ... > QjSimple: pjsip_endpt_register_module succeeded ... > QjSimple: outboundUri=sip:ps.sip.uniza.sk;lr;transport=udp > QjSimple: publish_enabled=FALSE > QjSimple: pjsua_acc_add succeeded ... > QjSimple: finished intializing SIP stack, start pjsua ... > QjSimple: initializing SIP stack ... done > QjSimple: Info: on_reg_state_slot called > QjSimple: Info: on_reg_state_slot: status=502 > QjSimple: Info: on_reg_state_slot: statustext=Bad Gateway > > When I setup as the transport protocol UDP6 there is no actions at all, > no dns, no sip, he re is debug log > > QjSimple: starting registration ... > QjSimple: initializing SIP stack ... > QjSimple: pjsua_create succeeded ... > QjSimple: "qjsimple" memory pool created ... > QjSimple: DLL loaded ... > QjSimple: GetNetworkParams found ... > QjSimple: GetNetworkParams done ... > QjSimple: cfg.nameserver_count=0 > QjSimple: pjsua_init succeeded ... > QjSimple: pjsip_endpt_register_module succeeded ... > QjSimple: adding UDP6 transport ... > QjSimple: pjsip_endpt_register_module succeeded ... > QjSimple: outboundUri=sip:ps.sip.uniza.sk;lr;transport=udp > QjSimple: publish_enabled=FALSE > QjSimple: pjsua_acc_add succeeded ... > QjSimple: finished intializing SIP stack, start pjsua ... > QjSimple: initializing SIP stack ... done > > regards > > pavel > > > -----Original Message----- > > From: sr-users-bounces(a)lists.sip-router.org [mailto:sr-users- > > bounces(a)lists.sip-router.org] On Behalf Of Klaus Darilion > > Sent: Wednesday, April 20, 2011 12:49 PM > > To: sr-users(a)lists.sip-router.org > > Subject: Re: [SR-Users] Ipv6 SIP UA > > > > > > > > Am 20.04.2011 11:01, schrieb Pavel Segeč: > > > Hi, > > > > > > yes I can agree now, linphone is working well, qjsimple has some > > problem > > > > If you find some problems (except that it does not work over TCL/TLS: > > http://trac.pjsip.org/repos/ticket/1127) please let me know. > > > > klaus > > > > _______________________________________________ > > 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 > > > > > > __________ Informacia od ESET NOD32 Antivirus, verzia databazy 6057 > > (20110420) __________ > > > > Tuto spravu preveril ESET NOD32 Antivirus. > > > > http://www.eset.sk > > > > > > __________ Informacia od ESET NOD32 Antivirus, verzia databazy 6059 > > (20110420) __________ > > > > Tuto spravu preveril ESET NOD32 Antivirus. > > > > http://www.eset.sk > > > > > __________ Informacia od ESET NOD32 Antivirus, verzia databazy 6059 > (20110420) __________ > > Tuto spravu preveril ESET NOD32 Antivirus. > > http://www.eset.sk > > > > _______________________________________________ > 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 > > > __________ Informacia od ESET NOD32 Antivirus, verzia databazy 6059 > (20110420) __________ > > Tuto spravu preveril ESET NOD32 Antivirus. > > http://www.eset.sk > __________ Informacia od ESET NOD32 Antivirus, verzia databazy 6059 (20110420) __________ Tuto spravu preveril ESET NOD32 Antivirus. http://www.eset.sk

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Re: [SR-Users] Ipv6 SIP UA
by Klaus Darilion 21 Apr '11

21 Apr '11

Am 20.04.2011 11:01, schrieb Pavel Segeč: > Hi, > > yes I can agree now, linphone is working well, qjsimple has some problem If you find some problems (except that it does not work over TCL/TLS: http://trac.pjsip.org/repos/ticket/1127) please let me know. klaus

2 1

xmlrpc requests counted as unsupported_methods
by Juha Heinanen 21 Apr '11

21 Apr '11

looks like core stats counts xmlrpc requests served by route [xmlrpc_requests] as unsupported_methods. is there anything a user can do about it or is it a bug that should be fixed? -- juha

2 4

Broken Via/reply-matching for natping OPTION
by Andreas Granig 20 Apr '11

20 Apr '11

Hi, When kamailio-3.1 sends OPTIONS for natping, the Via branch is wrong, e.g: OPTIONS sip:192.168.51.1:5060 SIP/2.0. Via: SIP/2.0/UDP 127.0.0.1:5062;branch=0. Route: <something> Should be branch=z9hG4bK<something> instead of branch=0, right? When a reply comes back, like this... SIP/2.0 100 Trying. Via: SIP/2.0/UDP 127.0.0.1:5062;branch=0;rport=5062. From: sip:pinger@<something> ... then kamailio logs this: INFO: <core> [forward.c:786]: broken reply to forward - no 2nd via The full debug log looks like this: DEBUG: <core> [parser_helpers.c:196]: path for branch: '<sip:lb@127.0.0.1:5060;lr;received=sip:192.168.51.1:5060>' DEBUG: <core> [parser/msg_parser.c:640]: SIP Reply (status): DEBUG: <core> [parser/msg_parser.c:642]: version: <SIP/2.0> DEBUG: <core> [parser/msg_parser.c:644]: status: <100> DEBUG: <core> [parser/msg_parser.c:646]: reason: <Trying> DEBUG: <core> [parser/parse_via.c:1287]: Found param type 232, <branch> = <0>; state=6 DEBUG: <core> [parser/parse_via.c:1287]: Found param type 235, <rport> = <5062>; state=16 DEBUG: <core> [parser/parse_via.c:2300]: end of header reached, state=5 DEBUG: <core> [parser/msg_parser.c:515]: parse_headers: Via found, flags=2 DEBUG: <core> [parser/msg_parser.c:517]: parse_headers: this is the first via DEBUG: <core> [receive.c:145]: After parse_msg... DEBUG: tm [t_lookup.c:1081]: DEBUG: t_check_msg: msg id=1 global id=0 T start=0xffffffffffffffff DEBUG: <core> [parser/parse_to.c:803]: end of header reached, state=9 DEBUG: <core> [parser/msg_parser.c:187]: DEBUG: get_hdr_field: <To> [23]; uri=[sip:192.168.51.1:5060] DEBUG: <core> [parser/msg_parser.c:189]: DEBUG: to body [sip:192.168.51.1:5060#015#012] DEBUG: <core> [parser/msg_parser.c:167]: get_hdr_field: cseq <CSeq>: <1> <OPTIONS> DEBUG: tm [t_lookup.c:1055]: DEBUG: t_reply_matching: failure to match a transaction DEBUG: tm [t_lookup.c:1150]: DEBUG: t_check_msg: msg id=1 global id=1 T end=(nil) DEBUG: <core> [parser/msg_parser.c:201]: DEBUG: get_hdr_body : content_length=0 DEBUG: <core> [parser/msg_parser.c:103]: found end of header INFO: <core> [forward.c:786]: broken reply to forward - no 2nd via DEBUG: <core> [usr_avp.c:646]: DEBUG:destroy_avp_list: destroying list (nil) Thanks, Andreas

4 6

planning next major release 3.2
by Daniel-Constantin Mierla 20 Apr '11

20 Apr '11

Hello, soon we should sketch the roadmap for the next major release 3.2. There are some private git branches with lot of work on it, so we need to synchronize a bit and decide about the proper time: before or after summer. Several devels will be in Berlin for Linuxtag, but no so much overlapping for a big face to face meeting, therefore might be better to have a IRC chat conference sometime in the near future. Next week will be fine for me, say Wednesaday till Friday, in the afternoon to allow fair time of the day for US timezones. If you have other proposals, speak up. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

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Re: [SR-Users] Ipv6 SIP UA
by Daniel-Constantin Mierla 20 Apr '11

20 Apr '11

Hello, On 4/19/11 9:58 AM, Pavel Segeč wrote: > Hi, > > may someone help me and recommend a software SIP client with a good IPv6 > support implemented? I tried Xlite, Bria, Linphone, SIP communicator and > Jitsi but with a bad results. Each have some problem within pure IPv6 > environment. the last sipit I attended (last year, though) was not rich at all in IPv6 softphones. In the past there was a patched version of KPhone we used at FhG Fokus for some tests, but I wouldn't call it production ready: http://old.iptel.org/products/kphone/ From what was listed above, you touched most of the active softphones, afaik ekiga,qutecom and twinkle do not have it, not sure about zoiper. Cheers, Daniel

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