September 2010 - sr-users

Kamailio --> Mitel (Not Found) ¿INVITE ISSUES?
by Tincho ylm 27 Sep '10

27 Sep '10

Hi all! I really don't know why "Mitel" rejects my calls. I'm using Kamailio to forward calls to Mitel. A little more graphic: Please see the picture: http://s3.subirimagenes.com:81/otros/5226539form.jpg SIP PHONE (Linksys) ---> Kamailio (1.5.4) ----> Mitel ----> Mitel Phone Mitel rejects my calls with "404 Not Found". Ok, you may think: "the extension that you are calling doesn't exists".. please dont think that. (One more thing: If I try to make the same scene using Asterisk instead Kamailio everything works fine.) So, I made a sip capture to see what happens: Sip Phone -> 100 192.168.10.140 -> Sip Phone 192.168.10.150 -> Kamailio 192.168.10.160 -> Mitel Mitel Phone -> 200 Kamailio U 192.168.10.140:5060 -> 192.168.10.150:5060 INVITE sip:200@192.168.10.150 <sip%3A200(a)192.168.10.150> SIP/2.0. Via: SIP/2.0/UDP 192.168.10.140:5060;branch=z9hG4bK-d063d53a. From: "Sip Phone" <sip:100@192.168.10.150 <sip%3A100(a)192.168.10.150> >;tag=d396005aaf3ab9a2o0. To: "Mitel Phone" <sip:200@192.168.10.150 <sip%3A200(a)192.168.10.150>>. Call-ID: d4de30ba-eb6944c2 [!at] 192.168.10.140 (replace the [!at] with a @). CSeq: 101 INVITE. Max-Forwards: 70. Contact: "Sip Phone" <sip:100@192.168.10.140:5060>. Expires: 240. User-Agent: Linksys/SPA941-5.1.8. Content-Length: 395. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. Supported: replaces. Content-Type: application/sdp. U 192.168.10.150:5060 -> 192.168.10.140:5060 SIP/2.0 100 Giving a try. Via: SIP/2.0/UDP 192.168.10.140:5060 ;branch=z9hG4bK-d063d53a;rport=5060;received=192.168.10.140. From: "Sip Phone" <sip:100@192.168.10.150 <sip%3A100(a)192.168.10.150> >;tag=d396005aaf3ab9a2o0. To: "Mitel Phone" <sip:200@192.168.10.150 <sip%3A200(a)192.168.10.150>>. Call-ID: d4de30ba-eb6944c2 [!at] 192.168.10.140 (replace the [!at] with a @). CSeq: 101 INVITE. Server: Kamailio (1.5.4-notls (i386/linux)). Content-Length: 0. U 192.168.10.150:5060 -> 192.168.10.160:5060 INVITE sip:200@192.168.10.150 <sip%3A200(a)192.168.10.150> SIP/2.0. Via: SIP/2.0/UDP 192.168.10.150;branch=z9hG4bKc17.8e746ba.0. Via: SIP/2.0/UDP 192.168.10.140:5060 ;rport=5060;received=192.168.10.140;branch=z9hG4bK-d063d53a. From: "Sip Phone" <sip:100@192.168.10.150 <sip%3A100(a)192.168.10.150> >;tag=d396005aaf3ab9a2o0. To: "Mitel Phone" <sip:200@192.168.10.150 <sip%3A200(a)192.168.10.150>>. Call-ID: d4de30ba-eb6944c2 [!at] 192.168.10.140 (replace the [!at] with a @). CSeq: 101 INVITE. Max-Forwards: 69. Contact: "Sip Phone" <sip:100@192.168.10.140:5060>. Expires: 240. User-Agent: Linksys/SPA941-5.1.8. Content-Length: 395. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. Supported: replaces. Content-Type: application/sdp. U 192.168.10.160:5060 -> 192.168.10.150:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 192.168.10.150;branch=z9hG4bKc17.8e746ba.0,SIP/2.0/UDP 192.168.10.140:5060 ;rport=5060;received=192.168.10.140;branch=z9hG4bK-d063d53a. From: "Sip Phone" <sip:100@192.168.10.150 <sip%3A100(a)192.168.10.150> >;tag=d396005aaf3ab9a2o0. To: "Mitel Phone" <sip:200@192.168.10.150 <sip%3A200(a)192.168.10.150> >;tag=0_4044193584-65506210. Call-ID: d4de30ba-eb6944c2 [!at] 192.168.10.140 (replace the [!at] with a @). CSeq: 101 INVITE. Content-Length: 0. U 192.168.10.160:5060 -> 192.168.10.150:5060 SIP/2.0 404 Not Found. Via: SIP/2.0/UDP 192.168.10.150;branch=z9hG4bKc17.8e746ba.0,SIP/2.0/UDP 192.168.10.140:5060 ;rport=5060;received=192.168.10.140;branch=z9hG4bK-d063d53a. From: "Sip Phone" <sip:100@192.168.10.150 <sip%3A100(a)192.168.10.150> >;tag=d396005aaf3ab9a2o0. To: "Mitel Phone" <sip:200@192.168.10.150 <sip%3A200(a)192.168.10.150> >;tag=0_4044193584-65506210. Call-ID: d4de30ba-eb6944c2 [!at] 192.168.10.140 (replace the [!at] with a @). CSeq: 101 INVITE. Contact: <sip:192.168.10.160>. Content-Length: 0. This is my Kamailio code from reenvites.. route[4] { t_relay("udp:192.168.10.160:5060"); t_on_reply("1"); exit; } If you pay attention to INVITES (Kamailio SIP messages) you will see: From: "Sip Phone" <sip:100@192.168.10.150 <sip%3A100(a)192.168.10.150> >;tag=d396005aaf3ab9a2o0. To: "Mitel Phone" <sip:200@192.168.10.150 <sip%3A200(a)192.168.10.150>>. I think that should be: From: "Sip Phone" <sip:100@192.168.10.150 <sip%3A100(a)192.168.10.150> >;tag=d396005aaf3ab9a2o0. To: "Mitel Phone" <sip:200@192.168.10.160 <sip%3A200(a)192.168.10.160>>. It could be the reason for Mitel rejects? Can I fix it? I can use TEXTOPS but I cant understand why Mitel rejects the Kamailio INVITES. I will thanks any help!

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Re: [SR-Users] xcap server modules !
by Daniel-Constantin Mierla 27 Sep '10

27 Sep '10

Hello, please use sr-users(a)lists.sip-router.org for questions about kamailio and ser On 9/24/10 12:17 PM, usman aftab wrote: > > I haven't been able to found these modules xhttp.so & xcap_server.so > mentioned on this link > http://sip-router.org/docbook/sip-router/branch/master/modules_k/xcap_serve… > > I am trying to setup an presence server with buddy list on kamailio > 3.0.3 on debian lenny. the xcap_server module is available only in current development version, which is planned to be released as 3.1.0 in few weeks. Backporting to 3.0.x may require some work. Therefore I advise you install the master branch and test it. I tested with SIP Communicator, Zoiper, a.s.o. and seems to work fine. Here are some guidelines: git clone --depth 1 git://git.sip-router.org/sip-router kamailio cd kamailio make FLAVOUR=kamailio include_modules="db_mysql presence presence_xml xcap_server" cfg make all make install The add to your config file the parts for xcap server, like in module readme (the link you pasted above). > If for any reason I need OpenXCAP. No, you don't need anything else, xcap_server module implements a XCAP server :-), so you have all-in-one and easy solution for SIMPLE presence. Thanks, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

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SEMS 1.3.0 released
by Stefan Sayer 27 Sep '10

27 Sep '10

SEMS 1.3.0 release With the release 1.3.0 the SIP Express Media Server continues to be the VoIP media and application server of choice for high volume VoIP services and flexible application development. This release focuses mostly on signaling, app development and interfacing. 100rel (PRACK) support, DNS cache and Session Timers B2BUA are a few of the additions there, and performance has been greatly improved especially for high volume signaling-only applications. The DSM state charts scripting engine has matured and can now be considered a viable and proven platform for rapid VoIP services development, one that does not sacrifice fine-grained control and flexibility with its easy- to-use high level abstraction of call and media control. Changelog: http://ftp.iptel.org/pub/sems/1.3/1.3.0/CHANGELOG Source: http://ftp.iptel.org/pub/sems/sems-1.3.0.tar.gz Packages: http://ftp.iptel.org/pub/sems/1.3/1.3.0/packages/ Most notable changes: * 100rel (PRACK) support * DNS cache, lb on SRV records * B2B with Session Timer * Signaling thread-pool support * json-rpc v2 module * SIP stack moved into core * optimizations, especially for signaling * many DSM improvements For a more complete list of changes, see doc/CHANGELOG. Users who upgrade from 1.2.x should be aware that the SIP stack settings have moved from sipctrl.conf into sems.conf, and that now both sip_ip and media_ip needs to be set, if it needs to be configured in your setup (i.e. they are not the first interface); furthermore, 100rel (PRACK) is supported by default. Contributors to this release were: Stefan Sayer, Raphael Coeffic, Bogdan Pintea, Ondrej Martinek, Peter Lemenkov, Andrey Samusenko, Robert Szokovacs, Juha Heinanen, Mark Sidell, Matthew Williams and Anton Zagorskiy. Many thanks to TelTech Systems Inc for sponsoring parts of the development that went into this release, and to the SEMS users for patches, bug reports and useful comments. Module: sems Branch: refs/tags/1.3.0 Tag: c788a58deb80d7ecc8d1e9c4be44e7ebfa307b0a URL: http://git.sip-router.org/cgi-bin/gitweb.cgi/sems/?a=tag;h=c788a58deb80d7ec… Apologies if you have received multiple copies of this message. With Best Regards Stefan Sayer -- Stefan Sayer VoIP Services Consulting and Development Warschauer Str. 24 10243 Berlin tel:+491621366449 sip:sayer@iptel.org email/xmpp:stefan.sayer@gmail.com

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releasing v3.1.0 on Oct 6
by Daniel-Constantin Mierla 27 Sep '10

27 Sep '10

Hello, I am proposing to release the next major version in about one week and a half, Wednesday, Oct 6. Stability of new features looks very good, it was an extended tresting period since couple of weeks overlapped yearly common summer holidays. Now we need to put a bit of efforts in migration documentation. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

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Re: [SR-Users] xcap server modules !
by Daniel-Constantin Mierla 24 Sep '10

24 Sep '10

private messages (sometime apart first ones) are discarded. Again, please keep the mailing list cc-ed, so other can learn from the conversation. Embedded xcap server works pretty well, I just sent an email for those that want to test it at voipuser.org: http://lists.sip-router.org/pipermail/sr-users/2010-September/065491.html 3.1 is in testing phase for a month by now, so is pretty stable. If you look to build a product to be ready in couple of weeks, then it is the best to start with this version, it really has lot of new cool features, you can find a summary at: http://sip-router.org/wiki/features/new-in-devel Thanks, Daniel On 9/24/10 1:40 PM, usman aftab wrote: > Thanks for the prompt reply. > Saved me a lot of time that I would have wasted trying to make it work > (Was about to build live distro out of it). > Much appreciated. > Thanx. > - > > On Fri, Sep 24, 2010 at 6:33 AM, Daniel-Constantin Mierla > <miconda(a)gmail.com <mailto:miconda@gmail.com>> wrote: > > Hello, > > please use sr-users(a)lists.sip-router.org > <mailto:sr-users@lists.sip-router.org> for questions about > kamailio and ser > > > On 9/24/10 12:17 PM, usman aftab wrote: > > > I haven't been able to found these modules xhttp.so & > xcap_server.so mentioned on this link > http://sip-router.org/docbook/sip-router/branch/master/modules_k/xcap_serve… > > I am trying to setup an presence server with buddy list on > kamailio 3.0.3 on debian lenny. > > > the xcap_server module is available only in current development > version, which is planned to be released as 3.1.0 in few weeks. > Backporting to 3.0.x may require some work. > > Therefore I advise you install the master branch and test it. I > tested with SIP Communicator, Zoiper, a.s.o. and seems to work fine. > > Here are some guidelines: > > git clone --depth 1 git://git.sip-router.org/sip-router > <http://git.sip-router.org/sip-router> kamailio > cd kamailio > make FLAVOUR=kamailio include_modules="db_mysql presence > presence_xml xcap_server" cfg > make all > make install > > The add to your config file the parts for xcap server, like in > module readme (the link you pasted above). > > > > If for any reason I need OpenXCAP. > > No, you don't need anything else, xcap_server module implements a > XCAP server :-), so you have all-in-one and easy solution for > SIMPLE presence. > > Thanks, > Daniel > > -- > Daniel-Constantin Mierla > http://www.asipto.com > > -- Daniel-Constantin Mierla http://www.asipto.com

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help testing 3.1
by Daniel-Constantin Mierla 24 Sep '10

24 Sep '10

Hello, before any major release of kamailio (openser), we install and run it live on voipuser.org sip service. It runs there for more than two weeks now, it even survived a 18 hours scan attack (approx 12.5 mio requests "with love from Latvia" - I will write an article about it), so it looks very solid. However, there are two features were enabled for this version and are not really heavily used yet: tls and presence/xcap. So if you don't plan to test by yourself a local instance of version 3.1 and have time, then maybe you can help by registering some phones at voipuser.org and play a bit there -- it will be very appreciated to report if works for you. If you dont have an account there or don't know about it, some details here: VoIPUser.org is hosting also our kamailio (openser) forum for many many years now, the service is open and free, when you register at http://www.voipuser.org for the forum, you get a sip account as well (there are some goodies included, like incoming UK did and INUM). For tls: - just configure your phone to use tls - if you need it, the port for tls is 5061. For example, with snom 370 you have to set the outbound proxy address to: sip.voipuser.org;transport=tls. For presence/xcap: - you can test if you have clients supporting SIMPLE presence in Presence Agent mode (not end-to-end mode) - the embedded xcap server is enabled, you have to set the xcap root url to http://sip.voipuser.org:5060/xcap-root or https://sip.voipuser.org:5061/xcap-root . Authentication for xcap service is done using same sip username and password Thanks, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

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Help regarding SIREMIS
by Zuhair Raza 24 Sep '10

24 Sep '10

HI I am newbie, I have a centos pc with httpd, php and mysql installed in it i followed the procedure at http://siremis.asipto.com/install/ when i m trying to browse my siremis web it is unable to submit my request... may be i am missing something configuring Apache please guide me how can i get out of it ? i have httpd 2 installed in my box thanks

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Memory leak
by Santiago Soares 23 Sep '10

23 Sep '10

Hello, We use Kamailio as a stateless load balancer. It runs on a VM, with 512 MB RAM, and it seems to have some kind of memory leak. The kamailio processes consume all available memory, until Kamailio crashes and restart. This occurs about once a day. Since it's stateless, and the restart is very fast, it's not a big deal, but I'd like to fix it. The strange thing is that I updated kamailio from 1.5.3 to 3.0.3 and the problem still ocurrs. I followed the instructions on http://www.kamailio.org/dokuwiki/doku.php/troubleshooting:memory in order to determine the cause, and stopped the traffic on the proxy. The allocated memory were not freed, after 1 hour the amount of free memory is still the same. Anybody has an ideia about what's happening, or how can I debug it? Thank you! Santiago Soares

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Kamailio 1.5.4 - Pike module ignoring remove_latancy parameter
by Miguel Baptista 23 Sep '10

23 Sep '10

Hi All, I'm running kamailio-1.5.4-tls and I want to enable pike module in it. I did some test but it isn't working properly. I mean it isn't acting according to the /remove_latancy/ parameter. When an IP address is blocked (cause it triggered the pike module), it should be blocked for the amount of time (seconds I presumed) defined on the /remove_latancy /parameter, right? but it isn't Here is my pike module config (it's just a test config) # ---- Pike --- /* we are usign default values. We should tunning it up */ modparam("pike", "sampling_time_unit", 30) modparam("pike", "reqs_density_per_unit", 10) modparam("pike", "remove_latency", 3600) modparam("pike", "pike_log_level",-1) and here is the output /Sep 10 *13:33:35* sip /home/kamailio/sbin/kamailio[21414]: PIKE - BLOCKing ip XXX.XXX.XXX.XXX, node=0xb5a2eb58 Sep 10 13:33:35 sip /home/kamailio/sbin/kamailio[21414]: Warning: too many requests from XXX.XXX.XXX.XXX:5060 Sep 10 *13:34:12* sip /home/kamailio/sbin/kamailio[21420]: PIKE - UNBLOCKing node 0xb5a2eb58 ... Sep 10 *13:35:22* sip /home/kamailio/sbin/kamailio[21418]: PIKE - BLOCKing ip XXX.XXX.XXX.XXX,, node=0xb5a2eb58 Sep 10 13:35:22 sip /home/kamailio/sbin/kamailio[21418]: Warning: too many requests from XXX.XXX.XXX.XXX,:5060 Sep 10 *13:36:12* sip /home/kamailio/sbin/kamailio[21420]: PIKE - UNBLOCKing node 0xb5a2eb58 / Shouldn't it be blocked for 3600 seconds? Then I changed the /remove_latancy/ parameter to /modparam("pike", "remove_latency", 334500) / / Sep 10 *13:37:09* sip /home/kamailio/sbin/kamailio[21462]: PIKE - BLOCKing ip XXX.XXX.XXX.XXX,, node=0xb5986b90 Sep 10 13:37:09 sip /home/kamailio/sbin/kamailio[21462]: Warning: too many requests from XXX.XXX.XXX.XXX,:5060 Sep 10 *13:37:52* sip /home/kamailio/sbin/kamailio[21466]: PIKE - UNBLOCKing node 0xb5986b90 / but the it didn't seem to have any real difference. Any ideas? / / Best Regards, Miguel Baptista

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rtpproxy (k): removal of force_rtpproxy
by Ovidiu Sas 22 Sep '10

22 Sep '10

Hello all, I would like to propose the removal of force_rtp_proxy function from the rtpproxy (k) module. Instead, the rtpproxy_offer/rtpproxy_answer should be used. Why? - force_rtp_proxy no longer support 200ok/ACK SDP negotiation ('s' flag was deprecated); - all functionality of force_rtp_proxy is provided by rtpproxy_offer/rtpproxy_answer; - rtpproxy_offer/rtpproxy_answer is easier to use in the config. Objections? Regards, Ovidiu Sas

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