- sr-users - kamailio.org

LCR "weight" column question
by Ricardo Martinez 29 Sep '10

29 Sep '10

Hello. I’m wondering how the “Weight” column in the LCR ‘gw’ table Works. I have 5 gateways defined with different weights. gw_name grp_id ip_addr hostname port uri_scheme transport strip tag weight ping flags GW02 58 10.0.0.71 5060 1 1 4 0258 10 0 0 GW03 58 10.0.0.72 5060 1 1 4 0258 10 0 0 GW05 58 10.0.0.132 5060 1 1 4 0258 20 0 0 GW06 58 10.0.0.133 5060 1 1 4 0258 20 0 0 GW09 58 10.0.0.70 5060 1 1 4 0258 20 0 0 GW10 58 10.0.0.69 5060 1 1 4 0258 20 0 0 If I have 100 calls, I was expecting was to have 20 to GW05, 20 to GW06, 20 to GW09 and 20 to GW10 Finalle 10 to GW02 and 10 to GW02. Is this the correct interpretation of this parameter? Thanks in advance. Regards, Ricardo Martinez.-

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Problem with ser - INVITE size problem tcp->udp?
by Simone Felici 29 Sep '10

29 Sep '10

Hello to all! I need a little help with our ser installation (ser-2.0.0-rc1). The continuous groving up of our infrastructure and using even more codecs, causes the INVITE (udp) to be over 1500bytes. An external->incoming call to our proxy sip comes in with a size of ~1300 and will be forwarded to the end device (sua) over 1500. Ser is adding more than 200bytes in the udp packet. I can see 'INVITE' from ser to customer, but the INVITE is not answerred due the size. I'm sure about that, because in certain situations, deleting codec list the packet-size goes under the 1500bytes and all works. To main issue is from external calls to ser server to our customers. Calls started from customers to ser and then out are all ok. Now the question: Could the passage from UDP to TCP solve the issue? Could someone give me an example and how implement it into our ser configuration? The atteched configuration, ser on a mysql basis, is working well in expect the issue described. Thank's a lot for your help! Simon

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Dialog timeout route
by Alex Balashov 29 Sep '10

29 Sep '10

Greetings, Since the move to name-based routes in Kamailio v3.0, what implications does this have for the 'timeout_route' dialog context attribute? Has this been updated to reflect that change, or do I still have to use numerical routes for it? Can I now set this as a named constant, e.g. $dlg_ctx(timeout_route) = DIALOG_TIMEOUT_ROUTE; ... route[DIALOG_TIMEOUT_ROUTE] { ... } or is it a string? $dlg_ctx(timeout_route) = "DIALOG_TIMEOUT_ROUTE"; Thanks! -- Alex Balashov - Principal Evariste Systems LLC 1170 Peachtree Street 12th Floor, Suite 1200 Atlanta, GA 30309 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/

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sip-router 3.0.99 newbe question
by Rouskol Andrey 28 Sep '10

28 Sep '10

Hello, Could somebody check if default ser.cfg file is correct in the following section: route[AUTHENTICATION] { .... # check if the UID from the authentication meets the From header $authuid = $uid; if (!lookup_user("$fu.uid", "@from.uri")) { xlog("L_INFO","fu.uid lookup failed\n"); del_attr("$uid"); } if ($fu.uid != $fr.authuid) { sl_reply("403", "Fake Identity"); drop; } .. Because it didn't work for me till I've replaced: if (!lookup_user("$fu.uid", "@from.uri")) { with: if (!lookup_user("$fr.uid", "@from.uri")) { Best regards, Andrey

2 1

Siremis v2.0 preview
by Elena-Ramona Modroiu 28 Sep '10

28 Sep '10

Hi, Siremis version 2.0 is planned to work with upcoming Kamailio or SER 3.1 and by now is alpha version. If you want to have a preview, then you can access it at: http://demo.asipto.com/siremis20/ username: guest password: guest v2.0 brings a completely new face and many features, reusing the latest openbiz development framework, among them: - easy 4 step web-based installer - siremis account management (including roles per users) - customizable menu from web interface - customizable views from web interface - rework of many kamailio/ser related components such as subscriber, siptrace, generic and usrloc charts Siremis administration module, user profile management and web based customization elements are disabled in the demo, to avoid breaking the content. Packaging is still under work, but if someone is good at php/xml, likes to dig in by itself to sort out issues and wants to help with testing, I can ship a raw tarball, just drop me an email. Feedback about this new version is also appreciated, in order to improve it. Regards, Ramona

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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

1 0

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

2 1

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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missing rtpproxy documentatio from kamailio website
by Ovidiu Sas 22 Sep '10

22 Sep '10

The kamailio doc website doesn't have the new rtpproxy module documentation: http://www.kamailio.org/docs/modules/devel/ Also, ratelimit module is listed as being k and in fact it is generic now. Regards, Ovidiu Sas

2 2

mi ul_dump 'buffer too small'
by Juha Heinanen 22 Sep '10

22 Sep '10

is there anything one can do when 'mi ul_dump' gives error: 500 - Internal server error processing 's': buffer too small i guess there is too many contacts. -- juha

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Modifying $pU
by dotnetdub 22 Sep '10

22 Sep '10

Hi All I am receiving a Preferred-Identity from a customer which I need to modify (strip the first character a 0) and then send it as Remote Party ID Sending the RPID is no problem, my issue is stripping the character. I have tried a few things to do this. subst('/^P-Preferred-Identity:(.*)<sip:(0)(.*)@.*>(.*)$/P-Preferred-Identity:\1<sip:\3@domain >\4/ig') This substitution works fine but when I read $pU it still has the leading 0 I also looked at using re.subst on $pU but can't get it to work. What is the best way to achieve what I'm trying to do? Sorry for hassling you! Thanks, Brian

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[OT] Any utility to create cool SIP flows in HTML format?
by Iñaki Baz Castillo 21 Sep '10

21 Sep '10

Hi, I'm trying to create cool SIP flows to be displayed in a HTML. What I'm looking for is something like (or exactly) this: http://www.tech-invite.com/Ti-sip-dialog.html There you can click into messages to show the complete headers/body and so on. Does somebody know a tool to create such flows? Thanks for any suggestion. NOTE: There are some others utilities that allow creating TXT flows, as in a draft/RFC. I'm looking for something really more powerful than that. -- Iñaki Baz Castillo <ibc(a)aliax.net>

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Change in RTP Payload time on the fly
by Pranav Desai 20 Sep '10

20 Sep '10

Hello, We have a proxy server setup that runs Kamailiov3.0.1 + RTPProxyv1.2.1. Nathelper support has been enabled in Kamailio. On a certain route, we call force_rtp_proxy("z120"), i.e. with resizing parameter 'z' and argument 120ms. During a call where endpoints have negotiated the use of G729, RTPProxy sends packets with ptime=12 (120ms), however occasionally rtpproxy sends packets with ptime 6(60ms), 8(80ms), 4(40ms). Under what circumstance would RTPProxy send packets with ptime < 12 ? Regards, Pranav Desai

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Request for consulting assistance with SER scripts and setup
by Scott Seltzer 20 Sep '10

20 Sep '10

If there is anyone who would be able to provide assistance with SER programming, configuration changes, or migrations to SER3.0 please send email to: Scott(a)connectmevoice.com Thanks , Scott Seltzer

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Re: [SR-Users] Usrloc reload problem
by Jan Klepal 19 Sep '10

19 Sep '10

Hello Daniel, On Saturday September 18 2010 12:03:59 Daniel-Constantin Mierla wrote: > Hello, > > On 9/17/10 3:27 PM, Jan Klepal wrote: > > Hello, > > > > I run kamailio 3.0.3 with location table stored in database (PostgreSQL > > 8.4.2) with db_mode 2 (write-back) and upon kamailio restart some > > contacts from database are loaded incorrectly. Messages for those > > contacts are routed to invalid IP addresses. > > > > After some debugging I figured out that it happens only for contacts > > which "received" column value is null. For those contacts value is > > filled from previously loaded contact having non-null value in > > "received" column. > > > > it was a bug, hopefully fixed. Inside the postgres module, an internal > buffer was not reset. > > Test with master branch or take the patch from commit (it is small) and > do it with 3.0.x: > http://lists.sip-router.org/pipermail/sr-dev/2010-September/008994.html > > Since I have no pg server at hand, couldn't actually test it, so when > you report is working fine, I will backport to 3.0.x. > > Thanks, > Daniel Thank you for fast reply. Yes, this patch solved the problem. Contacts are loaded correctly from DB now. Kind Regards, Jan

2 1

Re: [SR-Users] Change in RTP Payload time on the fly
by Pranav Desai 19 Sep '10

19 Sep '10

Hello, Thank you for your comments. My answers are inline. > Hello, > > ptime and also the z flag are ms values, so a values of 6 or 8 ms are not > the recommended length of time, it should be an integer > multiple of 5 ms. I havn't yet a look the the source, but the nathelper > doc > say ZNN, maybe only the 12 is taken from your 120? The 'ZNN' in the documentation might be misleading, since I tried with 'Z120' and the variable 'requested_nsamples' is set to 960 samples (120/10*80). This variable is used by RTP resizer to resize the RTP packets. > But in fact the doc confuse with ZNN and an example of 100ms. Could you > try > again with 90? > I tried with ptime=80ms and ptime=60ms. But, I still have the similar results, i.e. I get occasional packets with ptime=40 and sometimes ptime=20ms. Regards, Pranav Desai > regards, > Andreas > > 2010/9/18 Pranav Desai <pdesai(a)signalogic.com> > >> >> Hello, >> >> We have a proxy server setup that runs Kamailiov3.0.1 + RTPProxyv1.2.1. >> Nathelper support has been enabled in Kamailio. >> >> On a certain route, we call force_rtp_proxy("z120"), i.e. with resizing >> parameter 'z' and argument 120ms. During a call where endpoints have >> negotiated the use of G729, RTPProxy sends packets with ptime=12 >> (120ms), >> however occasionally rtpproxy sends packets with ptime 6(60ms), 8(80ms), >> 4(40ms). >> >> Under what circumstance would RTPProxy send packets with ptime < 12 ? >> >> Regards, >> Pranav Desai >> >> _______________________________________________ >> 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

500 command not activated
by dotnetdub 18 Sep '10

18 Sep '10

Hi All, Kamailio 3.0.3 kamctl domain show 500 command not activated however I can run kamctl domain showdb which dumps out the domains from the database. What am I missing? Thanks Brian

2 5

Usrloc reload problem
by Jan Klepal 18 Sep '10

18 Sep '10

Hello, I run kamailio 3.0.3 with location table stored in database (PostgreSQL 8.4.2) with db_mode 2 (write-back) and upon kamailio restart some contacts from database are loaded incorrectly. Messages for those contacts are routed to invalid IP addresses. After some debugging I figured out that it happens only for contacts which "received" column value is null. For those contacts value is filled from previously loaded contact having non-null value in "received" column. Here is sample debug, which shows contact (row #97 in result set) with IP 10.3.71.1 is later used for contact without "received" value (row #172): db_postgres [km_dbase.c:149]: 0x888e00 PQsendQuery(select username,contact,expires,q,callid,cseq,flags,cflags,user_agent,received,path,socket,methods,last_modified from location ) ... db_postgres [km_res.c:242]: PQgetvalue(0x888e00,79,9)=[sip:10.3.71.1:5060] db_postgres [km_res.c:251]: [79][9] Column[received]=[sip:10.3.71.1:5060] ... db_postgres [km_res.c:242]: PQgetvalue(0x888e00,172,9)=[] ... db_postgres [km_res.c:242]: PQgetvalue(0x888e00,172,0)=[222333444] db_postgres [km_res.c:251]: [172][0] Column[username]=[222333444] db_postgres [km_res.c:242]: PQgetvalue(0x888e00,172,1)=[sip:222333444@195.70.1.200] db_postgres [km_res.c:251]: [172][1] Column[contact]=[sip:222333444@195.70.1.200] db_postgres [km_res.c:242]: PQgetvalue(0x888e00,172,2)=[2010-09-16 00:21:04] db_postgres [km_res.c:251]: [172][2] Column[expires]=[2010-09-16 00:21:04] db_postgres [km_res.c:242]: PQgetvalue(0x888e00,172,3)=[-1] db_postgres [km_res.c:251]: [172][3] Column[q]=[-1] db_postgres [km_res.c:242]: PQgetvalue(0x888e00,172,4)=[0fc6692b6570aaac04b3c85a3b9496c0(a)sipxsip.cz] db_postgres [km_res.c:251]: [172][4] Column[callid]=[0fc6692b6570aaac04b3c85a3b9496c0(a)sipxsip.cz] db_postgres [km_res.c:242]: PQgetvalue(0x888e00,172,5)=[397] db_postgres [km_res.c:251]: [172][5] Column[cseq]=[397] db_postgres [km_res.c:242]: PQgetvalue(0x888e00,172,6)=[0] db_postgres [km_res.c:251]: [172][6] Column[flags]=[0] db_postgres [km_res.c:242]: PQgetvalue(0x888e00,172,7)=[0] db_postgres [km_res.c:251]: [172][7] Column[cflags]=[0] db_postgres [km_res.c:242]: PQgetvalue(0x888e00,172,8)=[Asterisk SIPXSIP] db_postgres [km_res.c:251]: [172][8] Column[user_agent]=[Asterisk SIPXSIP] db_postgres [km_res.c:242]: PQgetvalue(0x888e00,172,9)=[] db_postgres [km_res.c:242]: PQgetvalue(0x888e00,172,10)=[] db_postgres [km_res.c:242]: PQgetvalue(0x888e00,172,11)=[udp:80.200.1.100:5060] db_postgres [km_res.c:251]: [172][11] Column[socket]=[udp:80.200.1.100:5060] db_postgres [km_res.c:242]: PQgetvalue(0x888e00,172,12)=[] db_postgres [km_res.c:242]: PQgetvalue(0x888e00,172,13)=[2010-09-16 00:11:04] db_postgres [km_res.c:251]: [172][13] Column[last_modified]=[2010-09-16 00:11:04] <core> [db_row.c:119]: allocate 448 bytes for row values at 0x998120 <core> [db_val.c:117]: converting STRING [222333444] <core> [db_val.c:127]: allocate 10 bytes memory for STRING at 0x944008 <core> [db_val.c:117]: converting STRING [sip:222333444@195.70.1.200] <core> [db_val.c:127]: allocate 27 bytes memory for STRING at 0x9378d8 <core> [db_val.c:157]: converting DATETIME [2010-09-16 00:21:04] <core> [db_val.c:106]: converting DOUBLE [-1] <core> [db_val.c:117]: converting STRING [0fc6692b6570aaac04b3c85a3b9496c0(a)sipxsip.cz] <core> [db_val.c:127]: allocate 44 bytes memory for STRING at 0x925cd8 <core> [db_val.c:73]: converting INT [397] <core> [db_val.c:73]: converting INT [0] <core> [db_val.c:73]: converting INT [0] <core> [db_val.c:117]: converting STRING [Asterisk SIPXSIP] <core> [db_val.c:127]: allocate 17 bytes memory for STRING at 0x9061e8 <core> [db_val.c:117]: converting STRING [sip:10.3.71.1:5060] <core> [db_val.c:127]: allocate 19 bytes memory for STRING at 0x905fc8 <core> [db_val.c:56]: converting NULL value <core> [db_val.c:117]: converting STRING [udp:80.200.1.100:5060] <core> [db_val.c:127]: allocate 22 bytes memory for STRING at 0x905da8 <core> [db_val.c:73]: converting INT [4767] <core> [db_val.c:157]: converting DATETIME [2010-09-16 00:11:04] To make things clear, all rows after row #97 had "received" value null, and row #172 was UA I ran tests on... in the other words contact with null "received" value is filled with previous non-null value as long next non-null row is fetched from DB and following null contacts are filled with its value and so on. When the contact having invalid "registered" re-register it will overwrite value in memory and calls are routed to proper IP address. This query is sent to database then: db_postgres [km_dbase.c:149]: 0x888e00 PQsendQuery(update location set expires='2010-09-16 00:30:58',q=-1.00 ,cseq=103,flags=0,cflags=0,user_agent='Asterisk SIPXSIP',received=NULL,path=NULL,socket='udp:80.200.1.100:5 060',methods=NULL,last_modified='2010-09-16 00:20:58' where username='222333444' AND contact='sip:222333444@195.70.1.200' AND callid='0fc6692b6570aaac04b3c85a3b9496c0(a)sipxsip.cz') I've also checked status of this record in usrloc by "kamctl ul show"... both before and after kamailio restart state was CS_SYNC. Is this bug or should I somehow force kamailio to store "received" into database? Kind Regards, Jan Klepal

2 1

Kamailio Jabberd Integration
by shoks 17 Sep '10

17 Sep '10

Hi, Has anyone managed to integrate Kamailio and Jabberd to support presence and messaging functions (XMPP). I've tried to setup based on notes and emails that I found online, but still could'nt make it work. I appreciate if someone can share their configuration or give me a link. Thanks, sho

3 4

strange acc behavior?
by Klaus Darilion 17 Sep '10

17 Sep '10

Hi! I use acc module with extra accounting. modparam("acc", "db_extra", "direction=$avp(direction)") I set $avp(direction) during request processing (in main request route). The value stored with INVITE and BYE is fine, but ACK requests do not have the assigned value, but the same value as the INVITE request. Is this by design? Is this a limitation of tm and AVPs? Thanks Klaus

3 4

Stripping variable amount of digits
by Iñaki Baz Castillo 16 Sep '10

16 Sep '10

Hi, I get a prefix string from a database and store it in $var(prefix). Such prefix can contain any number of digits (no fixed length). Now I must to check if the RURI username begins with that prefix and in that case strip it. I can get the first step (detect it) using pcre_match function: if pcre_match("$rU", "^$var(prefix)") { # Prefix match. But I cannot get the second part working (stripping it). I've tryed everything: - strip() is not valid as just allows an integer (no pv). - subst_user() doesn't accept pv: if subst_user("/^$var(prefix)(.*)$/\1/") Any ideas please? Thanks a lot. -- Iñaki Baz Castillo <ibc(a)aliax.net>

2 3

Doubd about Dialplan module
by rabs＠dimension-virtual.com 16 Sep '10

16 Sep '10

Hi all, I'm tryting to implement a E164 normalizer using the dialplan module, by now I have 3 simple rules: dpid priority matchop matchex matchlen subsex replaceex attributes 0 1 1 (00|\+)([1-9][0-9]+) 0 (00|\+)([1-9][0-9]+) \2 0 1 1 ([5-9][0-9]{8}) 0 ([5-9][0-9]{8}) 34\1 The first rule 'it's supposed' to remove + or 00 from full qualified E164 numbers, the seccond rule 'should' add 34 as prefix to all 'valid' spanish numbers, they was running on some little test I did, but when try to put them on pre-production enviroment I get strage results. I call dp_translate as dp_translate("0","$rU/$rU"); And this what I get : Input RURI Output RURI 900XXXXXX(a)dom.tld XXXXXX(a)dom.tld And the quetion is why?, are my match rules wrong? why subex and matchex must be the same? Best regards. -- Raul Alexis Betancor Santana Dimension Virtual S.L.

4 4

string manipulation from db
by anthony thomas 16 Sep '10

16 Sep '10

Hello list, I am afraid I am asking something stupid but I cannot find a way to do this cleanly, so here it goes.... I am retrieving values from a db using either carrierroute or mtree modules to send a 302 to my softswitch. It expects the message as: SIP/2.0 302 Moved Temporarily Via: SIP/2.0/UDP 134.111.48.66:5060;branch=zgfdgdfgf2491b5437 Contact: <sip:989897@X.X.X.X;rg=4003>;q=0.2 Contact: <sip:989897@X.X.X.X:rg=321>;q=0.8 To: <sip:989897@X.X.X.X:5066>;tag=1 ..... Where the important things are the rg: those are the trunks in the SS) and the q (probability) and there can be multiple rgs for the same prefix (2 in this example) My idea was to keep in the db something like this: Prefix:98 Route:4003,0.2|321,0.8 And then inject those values in the contact header, but I am having a hard time finding the right way to do it. Is there any way to do it without calling the perl module? Many thanks, A

4 3

About avp's, var's, null, 0 and empty string
by Iñaki Baz Castillo 16 Sep '10

16 Sep '10

Hi, I would like to summarize how $avp's and $var's handle the null value, 0 and empty string. For this I've writen the following test I've checked in Kamailio 1.5.4: -------------------------- $var(null) = null; $avp(s:null) = null; $var(empty) = ""; $avp(s:empty) = ""; xlog("var(null): '$var(null)'\n"); if $var(null) == null xlog("*** if var(null) == null \n"); if $var(null) xlog("*** if var(null) \n"); if $var(null) != null xlog("*** if var(null) != null \n"); if ! $var(null) xlog("*** if ! var(null) \n"); if $var(null) == '' xlog("*** if var(null) == '' \n"); if $var(null) == 0 xlog("*** if var(null) == 0 \n"); xlog("\n"); xlog("avp(s:null): '$avp(s:null)'\n"); if $avp(s:null) == null xlog("*** if avp(s:null) == null \n"); if $avp(s:null) xlog("*** if avp(s:null) \n"); if $avp(s:null) != null xlog("*** if avp(s:null) != null \n"); if ! $avp(s:null) xlog("*** if ! avp(s:null) \n"); if $avp(s:null) == '' xlog("*** if avp(s:null) == '' \n"); if $avp(s:null) == 0 xlog("*** if avp(s:null) == 0 \n"); xlog("\n"); xlog("var(empty): '$var(empty)'\n"); if $var(empty) == null xlog("*** if var(empty) == null \n"); if $var(empty) xlog("*** if var(empty) \n"); if $var(empty) != null xlog("*** if var(empty) != null \n"); if ! $var(empty) xlog("*** if ! var(empty) \n"); if $var(empty) == '' xlog("*** if var(empty) == '' \n"); if $var(empty)== 0 xlog("*** if var(empty) == 0 \n"); xlog("\n"); xlog("avp(s:empty): '$avp(s:empty)' \n"); if $avp(s:empty) == null xlog("*** if avp(s:empty) == null \n"); if $avp(s:empty) xlog("*** if avp(s:empty) \n"); if $avp(s:empty) != null xlog("*** if avp(s:empty) != null \n"); if ! $avp(s:empty) xlog("*** if ! avp(s:empty) \n"); if $avp(s:empty) == '' xlog("*** if avp(s:empty) == '' \n"); if $avp(s:empty) == 0 xlog("*** if avp(s:empty) == 0 \n"); ------------------------- The result is: ------------------------- var(null): '0' *** if var(null) != null *** if ! var(null) *** if var(null) == 0 avp(s:null): '<null>' *** if avp(s:null) == null *** if ! avp(s:null) var(empty): '' *** if var(empty) != null *** if ! var(empty) *** if var(empty) == '' CRITICAL:core:comp_scriptvar: invalid operation 20/3/4!! WARNING:core:do_action: error in expression (l=356) avp(s:empty): '' *** if avp(s:empty) != null *** if ! avp(s:empty) *** if avp(s:empty) == '' CRITICAL:core:comp_scriptvar: invalid operation 20/3/4!! WARNING:core:do_action: error in expression (l=372) ------------------------- Conclusions: - $var cannot store null value. If so it converts it to 0. - When cheking "if $avp(...) {" the result is false in case the content of the $avp is null, 0 or an empty string. - When cheking "if $var(...) {" the result is false in case the content of the $var is 0 or an empty string. Any thoughts on this? Regards. -- Iñaki Baz Castillo <ibc(a)aliax.net>

3 6

Calling force_rtp_proxy twice
by Ernest Mavrel 15 Sep '10

15 Sep '10

Hi I have problem with force_rtp_proxy. Kamailio server have multi network interfaces. I have two PSTN GW-s. And If is something wrong with first one, the call is send to the second via PSTN failover route. When I send call to the first GW I'm calling force_rtp_proxy("oc","10.10.10.1"); But the first GW respond with 404 Not found, so I send INVITE to the second GW with calling force_rtp_proxy("oc","10.10.20.1"); In SDP I see double IP's : o=root 26366 26367 IN IP4 10.10.10.110.10.20.1 s=session c=IN IP4 10.10.10.110.10.20.1 How to resolve that? Thank you Ernest

2 1

Calling force_rtp_proxy twice
by Ernest Mavrel 15 Sep '10

15 Sep '10

Hi I have problem with force_rtp_proxy. Kamailio server have multi network interfaces. I have two PSTN GW-s. And If is something wrong with first one, the call is send to the second via PSTN failover route. When I send call to the first GW I'm calling force_rtp_proxy("oc","10.10.10.1"); But the first GW respond with 404 Not found, so I send INVITE to the second GW with calling force_rtp_proxy("oc","10.10.20.1"); In SDP I see double IP's : o=root 26366 26367 IN IP4 10.10.10.110.10.20.1 s=session c=IN IP4 10.10.10.110.10.20.1 How to resolve that? Thank you Ernest

1 0

Re: [SR-Users] To Develop SIP Server
by Daniel-Constantin Mierla 14 Sep '10

14 Sep '10

please keep the list cc-ed. On 9/14/10 10:53 AM, Vijayalakshmi wrote: > Daniel, > Yes, I need to port SER to my server. I tried with liposip2. But > haven't got my result... have you tried to compile it? What is the C compiler of this OS? Does it have support for makefile system? Cheers, Daniel > > Regards. > Lakshmi Marudhanayagam > > Daniel-Constantin Mierla wrote: >> Hello, >> >> On 9/14/10 8:00 AM, Lakshmi Marudhanayagam wrote: >>> Hi, >>> >>> Iam trying to develop a SIP server in VxWorks. Please assist me >>> with how to >>> implement SIP in my Server. Where do I get these info...???? >>> >>> >> do you need to develop a new one (writing from scratch) or port >> kamailio/ser to this os? Have you tried to compile kamailio on it? >> >> Cheers, >> Daniel >> > -- Daniel-Constantin Mierla http://www.asipto.com

1 0

Siremis v1.0.1 released
by Elena-Ramona Modroiu 14 Sep '10

14 Sep '10

Hi, Siremis, the web management interface for Kamailio, released v1.0.1. It is a patch update with fixes since v1.0.0. More about it at: http://siremis.asipto.com/2010/09/14/siremis-v1-0-1-released/ The series 1.0.x will be kept compatible with Kamailio 3.0.x. A new major release is baked these days (new UI and features) to work with upcoming 3.1.0. Regards, Ramona -- http://www.asipto.com

1 0

pdb and carrier id zero in finnish LNP
by Mikko Lehto 14 Sep '10

14 Sep '10

Hi I am exploring the applicability of pdb module in finnish number portability scheme. Looks like carrier id 000 cannot be used as zero is already reserved for "the number could not be found" result in pdb protocol. (Queries seem to work after I changed MIN_PDB_CARRIERID in common.h to 0, but unfortunately that configuration just does not make sense.) Is there any way out of this other than using internally different value for carrier 0? I rather use correct ids if somehow possible to avoid mapping logic in routing script. Note about help text consistency: When going through pdbt.c there is this comment above import_csv function: /* Read a csv list from the given file and build a dtree structure. Format of lines in csv file: "<number prefix>;<carrier id>". Format of carrier id: "D[0-9][0-9][0-9]". Returns the number of lines imported or -1 on error. */ Isn't the third line incorrect? Same thing also in help text output of pdbt -h. docs/data_format.txt mentions letter D only in front of carrier name file. What is the meaning of D anyway in this context? Here is my awk pipe contribution to scripts/get_carrier_names_finland.sh: curl -o - "http://www2.ficora.fi/numerointi/nure_numbering.asp?nums=tot&lang=en" | awk '/<tbody>/, /<\/tbody>/' | awk -F"</td" -v RS="</tr" '{ gsub(/.*>/,"",$1) gsub(/.*>/,"",$2); gsub(/ä/,"ä",$2); gsub(/Å/,"Å",$2); gsub(/ö/,"ö",$2); if ( $2 != "") { printf "D%.3d %s\n",$1,$2 } }' -- Mikko Lehto

2 3

Kamailio Documentation - A Volunteer
by Anthony Mark 14 Sep '10

14 Sep '10

Dear Daniel, With reference to your comment (in italics below) and request for a documentation "marshal", I humbly apply to be given the honor of contributing to that effect. "We would need a documentation marshall that should update the docs as something related is discussed on the lists. But none really volunteered to do it. For developers is not always obvious or easy (due to lack of time) to track the doc changes." - "Anyone stepping forward? " I fully understand your point and position in the light of the volume of work you have on your hands. It may interest you to know that i assisted with testing some versions and also put together the "How to... FreeSwitch on CentOS" documentation for ASTPP (www.astpp.org). As a reference, you may browse http://www.astpp.org/?q=node/267 and http://www.astpp.org/?q=node/268 ( node/267 and node/268 ) of astpp.org site. For sure, i understand that the Kamailio project is bigger and more complex than the ASTPP project i worked on; hence would require more of me. 　 Please advice if the Kamailio project still needs assistance with documentation. 　 Looking towards reading from you soon. 　 "Agatehk"

2 1

To Develop SIP Server
by Lakshmi Marudhanayagam 14 Sep '10

14 Sep '10

Hi, Iam trying to develop a SIP server in VxWorks. Please assist me with how to implement SIP in my Server. Where do I get these info...???? Thanks in Advance Lakshmi Marudhanayagam:confused: -- View this message in context: http://old.nabble.com/To-Develop-SIP-Server-tp29697687p29697687.html Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

2 1

To Develop SIP Server
by Vijayalakshmi 14 Sep '10

14 Sep '10

Hi, Iam trying to develop a SIP server in VxWorks. Please assist me with how to implement SIP in my Server. Where do I get these info...???? Thanks in Advance...!!!! Lakshmi Marudhanayagam

2 1

help with tls error :sslv3 alert bad certificate
by peter_green lion 13 Sep '10

13 Sep '10

hi all, i have configure tls support as this link: http://www.kamailio.org/docs/tls-devel.html#id2451496 and i add certificate to 3CX sip phone is "cacert.pem" but when i register sip phone, the log file in kamailio server is : Sep 9 15:13:36 appliance /usr/local/sbin/kamailio[2146]: ERROR: tls [tls_server.c:392]: SSL error:error:14094412:SSL routines:SSL3_READ_BYTES:sslv3 alert bad certificate my configure in kamailio.cfg as : modparam("tls", "tls_method", "TLSv1") modparam("tls", "tls_method", "SSLv23") modparam("tls", "certificate", "/usr/local/etc/kamailio//tls/user/user-cert.pem") modparam("tls", "private_key", "/usr/local/etc/kamailio//tls/user/user-privkey.pem") modparam("tls", "ca_list", "/usr/local/etc/kamailio//tls/user/user-calist.pem") modparam("tls", "verify_certificate",0 ) modparam("tls", "require_certificate",0 ) please suggest to fix this error. thanks and regards. Peter Green.

2 12

dispatcher problem
by Klaus Darilion 13 Sep '10

13 Sep '10

Hi! I have some problems with 3.0 dispatcher module: module configuration: # ----- dispatcher ----- /* enable failover support */ modparam("dispatcher", "flags", 2) modparam("dispatcher", "dst_avp", "$avp(i:271)") modparam("dispatcher", "grp_avp", "$avp(i:272)") modparam("dispatcher", "cnt_avp", "$avp(i:273)") If I understand correctly, then dispatcher should send OPTIONS once a destination is in "probe" mode. But this never happens, e.g. if I set "probe" mode manually, the destination is set to active again: # kamctl fifo ds_list SET_NO:: 1 SET:: 1 URI:: sip:gucci.labs.nic.at flag=A URI:: sip:nvst.labs.nic.at flag=A # kamctl fifo ds_set_state i 1 sip:gucci.labs.nic.at # kamctl fifo ds_list SET_NO:: 1 SET:: 1 URI:: sip:gucci.labs.nic.at flag=I URI:: sip:nvst.labs.nic.at flag=A # kamctl fifo ds_set_state p 1 sip:gucci.labs.nic.at # kamctl fifo ds_list SET_NO:: 1 SET:: 1 URI:: sip:gucci.labs.nic.at flag=A URI:: sip:nvst.labs.nic.at flag=A I also tried ds_probing_mode=1 which IIUC should send OPTIONS also to active destinations to find out if the destination is alive. But I never see any OPTIONS requests. Is there something I am doing completely wrong? thanks Klaus

5 8

Remving ISUB tag in FROM field
by Alex rsm 10 Sep '10

10 Sep '10

Hi, Our openSER proxy is receiving SIP messages from PBX devices. The PBX includes its extension in the FROM field of the SIP message. Is there any function in the openSER to strip off the PBX extension (anything after username and before @domain)? Is uac_replace_from( ) function is the right one? 3d:13h:41m:33s INVITE sip:888-555-1212@10.0.0.1;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.5.0.1;branch=z9hG4bKac1169001476 Max-Forwards: 70 From: <sip:8885551212;isub=37313132@10.5.0.1.>;tag=1c1168990861 To: <sip:8665551212@10.0.0.1;user=phone> Call-ID: 1168989994392010134133(a)10.0.0.1 CSeq: 1 INVITE Contact: <sip:4075857112@10.5.0.1> Supported: em,timer,replaces,path,resource-priority,sdp-anat Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE User-Agent: CS-Pelloni/v.5.60A.021.004 Content-Type: application/sdp Content-Length: 261Thanks, AR

2 1

final question about tls support in kamailio.
by peter_green lion 10 Sep '10

10 Sep '10

hi all, i have install and try to fix my error many time. and i cannot recognize my problem is. final i think maybe it about openssl. my version is : OpenSSL 0.9.8e-fips-rhel5 01 Jul 2008. so please help me to upgrade it. i try to down load and install another version, but when i rn command openssl version , it don`t change . please help me final time. thanks all so much. Peter Green.

1 0

Kamailio 3.03
by dotnetdub 09 Sep '10

09 Sep '10

Hi All, Just went from 3.0 to 3.0.3 and when I try to start I get: Not starting kamailio: invalid configuration file! 0(13110) ERROR: <core> [modparam.c:143]: set_mod_param_regex: parameter <radius_config> not found in module <acc> 0(13110) : <core> [cfg.y:3333]: parse error in config file //etc/kamailio/kamailio.cfg, line 146, column 74: Can't set module parameter 0(13110) ERROR: <core> [modparam.c:143]: set_mod_param_regex: parameter <radius_flag> not found in module <acc> 0(13110) : <core> [cfg.y:3333]: parse error in config file //etc/kamailio/kamailio.cfg, line 147, column 33: Can't set module parameter 0(13110) ERROR: <core> [modparam.c:143]: set_mod_param_regex: parameter <radius_missed_flag> not found in module <acc> 0(13110) : <core> [cfg.y:3333]: parse error in config file //etc/kamailio/kamailio.cfg, line 148, column 40: Can't set module parameter 0(13110) ERROR: <core> [modparam.c:143]: set_mod_param_regex: parameter <radius_extra> not found in module <acc> 0(13110) : <core> [cfg.y:3333]: parse error in config file //etc/kamailio/kamailio.cfg, line 163, column 43: Can't set module parameter ERROR: bad config file (4 errors) Has something changed in the radius accounting module? Thanks, Brian

3 5

Error when integrate xmpp module to Kamailio 3.0.1
by Quốc Huy Nguyễn 09 Sep '10

09 Sep '10

Hi all, I have some problem when integrate module xmpp to Kamailio 3.0.1. I have already searched on google and mailing-list but can't find the right way to fix it. I install Kamailio 3.0.1 on server 192.168.1.25 (kamailio1.com), Ejabberd on server 192.168.1.21 (ejabberd.com). Here the config of kamailio1 server: # ---- xmpp ---- modparam("xmpp", "gateway_domain", "kamailio1.com") modparam("xmpp", "xmpp_domain", "ejabberd.com") modparam("xmpp", "xmpp_host", "ejabberd.com") modparam("xmpp", "backend", "component") modparam("xmpp", "xmpp_password", "secret") modparam("xmpp", "domain_separator", "*") modparam("xmpp", "xmpp_port", 5347) ........ if (method == "MESSAGE") { log("*** xmpp-handled MESSAGE message.\n"); if (xmpp_send_message()) { sl_send_reply("200", "Accepted"); } else { sl_send_reply("404", "Not found"); } return; } log("*** xmpp: unhandled message type\n"); sl_send_reply("503", "Service unavailable"); return; Here the config of ejabberd server: {5347, ejabberd_service, [ {access, all}, {shaper_rule, fast}, {ip, {192, 168, 1, 21}}, {hosts, ["kamailio1.com"], [{password, "secret"}]} ]}, I create 1 user on kamailio1 server: 103(a)kamailio1.com And 1 user on ejabberd.com: 102(a)ejabberd.com Test: - Login with account 103(a)kamailio1.com on sip communicator. Add 102*ejabberd.com(a)kamailio1.com on the sip contact list. - Login with account 102(a)ejabberd.com on pidgin. Add 103*kamailio1.com(a)ejabberd.com on the buddy list. - When 103(a)kamailio1.com chat to 102*ejabbed.com(a)kamailio1.com, kamailio1 can't send MESSAGE to ejabberd.com, and here the log on kamailio1 server: /usr/local/sbin/kamailio[4470]: ERROR: xmpp [xmpp.c:362]: invalid content-type 0x300ff /usr/local/sbin/kamailio[4463]: ERROR: xmpp [xmpp.c:362]: invalid content-type 0x300ff /usr/local/sbin/kamailio[4468]: ERROR: xmpp [xmpp.c:362]: invalid content-type 0x300ff /usr/local/sbin/kamailio[4470]: ERROR: xmpp [xmpp.c:362]: invalid content-type 0x300ff /usr/local/sbin/kamailio[4468]: ERROR: xmpp [xmpp.c:362]: invalid content-type 0x300ff /usr/local/sbin/kamailio[4468]: ERROR: xmpp [xmpp.c:362]: invalid content-type 0x300ff /usr/local/sbin/kamailio[4470]: ERROR: xmpp [xmpp.c:362]: invalid content-type 0x300ff /usr/local/sbin/kamailio[4462]: ERROR: xmpp [xmpp.c:362]: invalid content-type 0x300ff I think I had some wrong or miss something in the configuration of kamailio and ejabberd, please help me to check it. P/S: I also sent the sip trace on the attached file. In the sip trace file, the ip of sip client is 192.168.1.65, and the ip of pidgin client is 192.168.1.11 Thanks and Best Regards, Huy Nguyen Nguyễn Quốc Huy Post & Telecommunications Institute and Technology branch Ho Chi Minh City -------------------------------------------------- Email: huy_quocnguyen(a)live.com Mobile: +84 906755226 Y!M: huy_quocnguyen1987

3 7

Siremis 1.0.0 bug in LCR configuration?
by klaus.lists＠inode.at 09 Sep '10

09 Sep '10

Hello list, does anybody have experience in configuring LCR (gw and lcr table) in the SQL database with siremis? I found a problem that inserting a gateway via the menu "LCR Gateways" is not possible without manual adaption in the database. It is no problem inserting LCR Rules; the entries are representing all columns of the SQL table "lcr". However, when I insert gateways that are linked over the so called 'group_id' to the LCR Rules I can not find any field in the WebGUI where I can put in a "lcr_id". And that is the problem. When insertion of rules and gateways is finished in the WebGUI I have to reload the lcr configuration e.g. with the MI command "lcr_reload". But this is not possible, because kamailio is displaying an error message, which looks like: 'ERROR: lcr [lcr_mod.c:1280]: gw grp_id <10> of prefix <2910> has no gateways' The reason of this error message is that the lcr_id of the table "gw" is set to 0 (hardcoded). So it does not match when the rules are inserted with another lcr_id (e.g. 1 instead of 0). => why is it possible to insert a free lcr_id in the menu "LCR Rules" but not in the menu "LCR Gateways"? If only lcr_id '0' is supported in general, then this id should be fix in the menu "LCR Rules", too. Am I wrong or does anybody else have an opinion to this topic? regards, Klaus

2 3

is db accounting non-blocking?
by Klaus Darilion 08 Sep '10

08 Sep '10

Hi! Is DB accounting (mysql) asynchronous? I.e. if non-local DB is slow - does this affect Kamailio's message routing? thanks Klaus

2 1

TM question
by Alex Balashov 08 Sep '10

08 Sep '10

Hi, I have a somewhat unusual log entry after some calls: Sep 3 09:12:23 kamgw01 /usr/local/sbin/kamailio[542]: WARNING: tm [t_lookup.c:1551]: WARNING: script writer didn't release transaction These calls fail in that they are processed instantly, but no time is ever given to receive a response from the upstream gateway. For instance, the above call's initial INVITE was received at 09:12:23, a final request URI was resolved, and then this message, all in < 1 second. Trace on upstream gateway shows that the initial request was never received from Kamailio. There is one additional aspect to how these calls are being processed in script that almost certainly has a bearing on this, though I don't know what it is: They are handled through a gateway route route called route[OUTBOUND-ROUTE-TRY] that is, under normal circumstances reflecting its name, initially called from the request route, and subsequently from a failure route meant to advance it. Normal operation mode is like this: route { ... route(OUTBOUND-ROUTE-TRY); } route[OUTBOUND-ROUTE-TRY] { if(dialog_count_of_vendor_gw < limit) { t_on_failure("OUTBOUND-ROUTE-FAILURE"); if(!t_relay()) { sl_reply_error(); } exit; } else { ... # Advance index of AVP array to next gateway, etc. # if(!t_lookup_request()) t_newtran(); route(OUTBOUND-ROUTE-TRY); exit; } } ... # Armed in OUTBOUND-ROUTE-TRY with t_on_failure(). failure_route[OUTBOUND-ROUTE-FAILURE] { ... # If we have more routes to retry, call # OUTBOUND-ROUTE-TRY again. route(OUTBOUND-ROUTE-TRY); } However, what is happening to cause this error is a little different from the usual scenario. On the initial attempt, the dialog limit check you see in OUTBOUND-ROUTE-TRY is failing, so the route adds a new transaction and calls itself again once. Now, the reason I am creating a new transaction is because in certain failure modes when a transaction was never created (like, if the call also fails on the next attempt), it is otherwise not possible to send a stateful failure reply to the originating endpoint from the failure route, which is a requirement (one cannot issue stateless replies from failure routes). So, because this routine is called both from the failure route and from the request route, I create a new transaction in order to make it possible to send stateful replies (t_reply()) from within this route regardless of the type of the parent route calling it, whether it is failure route or request route. So, again, what is happening in the scenario in which I get the above log message is: 1. INVITE is received 2. Master request route runs OUTBOUND-ROUTE-TRY request route 3. OUTBOUND-ROUTE-TRY checks gateway limit, finds it is exceeded 4. OUTBOUND-ROUTE-TRY checks request against existing transaction, and if not found, creates a new one with t_newtran() 5. OUTBOUND-ROUTE-TRY calls itself again. 6. OUTBOUND-ROUTE-TRY succeeds this time, but evidently, t_relay() fails with the above message, or above message appears sometime before t_relay() is called. Any insight is appreciated! Thanks, -- Alex Balashov - Principal Evariste Systems LLC 1170 Peachtree Street 12th Floor, Suite 1200 Atlanta, GA 30309 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/

2 5

offset exceeds message size crash
by federico cabiddu 08 Sep '10

08 Sep '10

Hi, I'm using kamailio 1.5.4-notls and I'm experimenting crashes when an UAC sends an INVITE with a content-length greater then the effective body length. The error messages written on the logs is: CRITICAL:core:del_lump: offset exceeds message size (1266 > 1161) aborting... and this is the backtrace of the generated core file: #0 0x00002ad718ab307b in raise () from /lib/libc.so.6 (gdb) bt #0 0x00002ad718ab307b in raise () from /lib/libc.so.6 #1 0x00002ad718ab484e in abort () from /lib/libc.so.6 #2 0x0000000000418f53 in del_lump (msg=0x66de00, offset=1266, len=12, type=HDR_OTHER_T) at data_lump.c:292 #3 0x00002ad71a8145ba in alter_mediaip (msg=0x66de00, body=<value optimized out>, oldip=0x7fff81fe6700, oldpf=<value optimized out>, newip=0x7fff81fe66e0, newpf=2, preserve=0) at nathelper.c:1857 #4 0x00002ad71a821a3a in force_rtp_proxy (msg=0x66de00, str1=<value optimized out>, str2=<value optimized out>, offer=<value optimized out>) at nathelper.c:2871 #5 0x00002ad71a8238df in rtpproxy_offer1_f (msg=0x66de00, str1=0x65f370 "cof", str2=<value optimized out>) at nathelper.c:2391 #6 0x000000000040cc5a in do_action (a=0x65f400, msg=0x66de00) at action.c:874 #7 0x000000000040f19f in run_action_list (a=<value optimized out>, msg=0x66de00) at action.c:145 #8 0x0000000000454155 in eval_expr (e=0x65f4d0, msg=0x66de00, val=0x0) at route.c:1171 #9 0x0000000000453bd7 in eval_expr (e=0x65f518, msg=0x66de00, val=0x0) at route.c:1488 #10 0x0000000000453b7f in eval_expr (e=0x65f560, msg=0x66de00, val=0x0) at route.c:1493 #11 0x000000000040c4c9 in do_action (a=0x65ffe8, msg=0x66de00) at action.c:729 #12 0x000000000040f19f in run_action_list (a=<value optimized out>, msg=0x66de00) at action.c:145 #13 0x000000000040dbc9 in do_action (a=0x660528, msg=0x66de00) at action.c:746 #14 0x000000000040f19f in run_action_list (a=<value optimized out>, msg=0x66de00) at action.c:145 #15 0x000000000040dbc9 in do_action (a=0x6606c8, msg=0x66de00) at action.c:746 #16 0x000000000040f19f in run_action_list (a=<value optimized out>, msg=0x66de00) at action.c:145 #17 0x000000000040dac5 in do_action (a=0x656790, msg=0x66de00) at action.c:120 #18 0x000000000040f19f in run_action_list (a=<value optimized out>, msg=0x66de00) at action.c:145 #19 0x000000000040dbc9 in do_action (a=0x656860, msg=0x66de00) at action.c:746 #20 0x000000000040f19f in run_action_list (a=<value optimized out>, msg=0x66de00) at action.c:145 #21 0x000000000040dac5 in do_action (a=0x6560b0, msg=0x66de00) at action.c:120 #22 0x000000000040f19f in run_action_list (a=<value optimized out>, msg=0x66de00) at action.c:145 #23 0x000000000040f4f3 in run_top_route (a=0x64b870, msg=0x66de00) at action.c:120 #24 0x0000000000444e90 in receive_msg ( buf=0x619a20 "INVITE sip:xxxxxxxx@xxxxxxxxxxxx SIP/2.0\r\nVia: SIP/2.0/UDP xxx.xxx.xxx.xxx:xxxx;branch=z9hG4bK-d8754z-24245342621eb55b-1---d8754z-;rport\r\nMax-Forwards: 69\r\nContact: <sip:xxxxxxxx@xxx.xxx.xxx.xxx"..., len=1161, rcv_info=0x7fff81fe86e0) at receive.c:175 #25 0x0000000000479254 in udp_rcv_loop () at udp_server.c:449 #26 0x0000000000427237 in main (argc=7, argv=0x7fff81fe88e8) at main.c:774 I couldn't get to reproduce this behavior in my test development (it has newer version of glibc) in which I only get the messages: ERROR:core:anchor_lump: offset exceeds message size (1125 > 714)... ERROR:nathelper:force_rtp_proxy: anchor_lump failed Looking into nathelper code, extract_body function I found that the body->len value is taken from Content-Length header, so i added the following piece of code: --- nhelpr_funcs.c.orig 2010-09-02 14:04:09.891649254 +0200 +++ nhelpr_funcs.c 2010-09-02 14:17:40.183747107 +0200 @@ -196,6 +196,12 @@ LM_ERR("message body has length zero\n"); goto error; } + + if (body->len + body->s > msg->buf + msg->len) { + LM_ERR("content-length exceeds packet-length by %d\n", + (body->len + body->s) - (msg->buf + msg->len)); + body->len=strlen(body->s); + } /* no need for parse_headers(msg, EOH), get_body will * parse everything */ This way if the Content-Length header is greater then the effective body length body->len is corrected with the real value. This solved for the moment, but I'm not sure if this is a good approach and I still don't understand why in the test platform I cannot reproduce the crash. Regards, Federico Cabiddu

2 2

route_graph tool
by Daniel-Constantin Mierla 07 Sep '10

07 Sep '10

1 0

time format in db_flatstore
by SIP Projects 07 Sep '10

07 Sep '10

Using version 1.5.2. Trying to migrate acc from db_mysql to db_flatstore to reduce io wait during heavy dialing. However, the default time format changes from normal YYYY-MM-DD hh:mm:ss to unixtime in flatstore. I tried using $TF but its format is even more non-standard. Is there a way to change time format for either the default acc time field in the db_flatstore module or $TF? Thanks, Matt

2 1

some issues in Kamailio WITH FreeSWITCH, SOS
by Abhishek [ABK] Kumar 07 Sep '10

07 Sep '10

Hi SiP-Users, I got kamailio installed and configured to work using mysql with FreeSWiTCH using this HowTo wiki freeswitch:kamailio-3.0.x-freeswitch-1.0.6d-ms [Asipto - SIP and VoIP Knowledge Base Site]<http://kb.asipto.com/freeswitch:kamailio-3.0.x-freeswitch-1.0.6d-ms>: http://kb.asipto.com/freeswitch:kamailio-3.0.x-freeswitch-1.0.6d-ms This is also reference on FreeSWITCH portal and was the only decent tutorial I was able to find on integrating Kamailio with FreeSWITCH. Followed everything successfully, but I have a slight confusion/problem in its wonderful working... >From all tests I performed, I perceived following activities: a.) it handled the extensions added to it very well, and carried over the call to destination extension b.) if destination extension is also added to it, call established without intervention of FreeSWITCH c.) if extension is not added to it but FreeSWITCH, the call reached FreeSWITCH but was rejected there d.) Red5Phone (developed over RED5 Server, @Google-Code) could dial with success but didn't received any Now, *for point (a)* Thank You! *for point (b)* it should be transferred to FreeSWITCH and that should be allowed to establish call, otherwise all dial-plans,etc. goes waste *for point (c)* may be some settings in Wiki are Obsolete OR New, could you please point it out *for point (d)* it may be due to point (b) -- -- Regards, Abhishek Kumar https://sites.google.com/site/abhikumar163/ -- --------------ABK-----mail.signature-------------------- <http://www.blogger.com/profile/06276198262605731980><http://abhishekkr.deviantart.com/><http://www.facebook.com/aBionic><http://www.twitter.com/aBionic><http://sourceforge.net/users/abhishekkr><http://www.youtube.com/user/1ABK><http://in.linkedin.com/in/abionic> ----------------------------------------------------------- ~=ABK=~

2 1

kamailio and asterisk proiblem
by chuks＠cybernergies.com 07 Sep '10

07 Sep '10

2 1

[patch] make install fails with a bash syntax error
by Dmitry Goncharov 07 Sep '10

07 Sep '10

Please, have a look http://sip-router.org/tracker/index.php?do=details&task_id=87. Regards, Dmitry

2 1

High performance routing options
by anthony thomas 07 Sep '10

07 Sep '10

Hi everyone, We are looking at offloading routing logic out of our existing softswitch using openser as an external redirect server. Our routing logic is not terribly complicated and I think both carrierroute and LCR modules would be able to handle it but we have a pretty large route list table with around 30 million records (our routing table is 300k codes but keep different tables per client), 25-30 calls per sec would be enough. >From a performance point of view, what would be the best bet within both modules? also, are both modules actively maintained? Thanks, Anthony

6 13

Routing issues with SIP Route headers.
by Siddhardha Garige 07 Sep '10

07 Sep '10

Hello all, I am having little trouble routing registrations based on route header in following configuration. UA sends registration to P1 and P1 is ignoring “route header” and forwards registration request to Registrar. I am using default kamailio configuration. Is there any thing I need to add to default config to enable route headers? UA----------------->proxy1(kamailio)------------------->proxy2------------------>Registrar REGISTER sip:Registrar.com SIP/2.0 Via: SIP/2.0/UDP 10.18.0.67:5060;branch=z9hG4bKd209323e;rport Route: <sip:proxy2.com;lr> Max-Forwards: 70 From: <sip:xxxxxx@registrar.com>;tag=a85f9424 To: <sip:xxxxx@registrar.com> Contact: <sip:xxxxxxx@10.18.0.67> Call-ID: 431368ae25a3e232 CSeq: 100 REGISTER Expires: 840 Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,UPDATE,SUBSCRIBE,NOTIFY Supported: User-Agent: xg-tx60 3.1.17 (VeriCall Edge) Content-Length: 0 ThanksSid "May the light be with you." ______________________________________________ Siddhardha Garige www.luminepixels.com

3 4

please help to register sip phone to kamailio server via tls support.
by peter_green lion 07 Sep '10

07 Sep '10

hi all, I have configured tls support in kamailio, but i cannot register sip phone. my configure : I create cert and private key as: "kamctl tls userCERT user" log show : Creating directory /usr/local/etc/kamailio//tls/user Creating user certificate request Generating a 512 bit RSA private key ..++++++++++++ ...................++++++++++++ writing new private key to '/usr/local/etc/kamailio//tls/user/user-privkey.pem' ----- Signing certificate request Using configuration from /usr/local/etc/kamailio//tls/request.conf Enter pass phrase for ./rootCA/private/cakey.pem: Check that the request matches the signature Signature ok The Subject's Distinguished Name is as follows commonName :PRINTABLE:'somename.somewhere.com' stateOrProvinceName :PRINTABLE:'Some State' countryName :PRINTABLE:'XY' emailAddress :IA5STRING:'root(a)somename.somewhere.com' organizationName :PRINTABLE:'My Large Organization Name' organizationalUnitName:PRINTABLE:'My Subunit of Large Organization' Certificate is to be certified until Sep 4 09:13:58 2011 GMT (365 days) Sign the certificate? [y/n]:y 1 out of 1 certificate requests certified, commit? [y/n]y Write out database with 1 new entries Data Base Updated Generating CA list DONE INFO: Private key is locate at /usr/local/etc/kamailio//tls/user/user-privkey.pem INFO: Certificate is locate at /usr/local/etc/kamailio//tls/user/user-cert.pem INFO: CA-List is locate at /usr/local/etc/kamailio//tls/user/user-calist.pem I add to kamailio.cfg enable_tls=1 tcp_async=no modparam("tls", "tls_method", "TLSv1") modparam("tls", "certificate", "/usr/local/etc/kamailio//tls/user/user-cert.pem") modparam("tls", "private_key", "/usr/local/etc/kamailio//tls/user/user-privkey.pem") modparam("tls", "ca_list", "/usr/local/etc/kamailio//tls/user/user-calist.pem") modparam("tls", "verify_certificate", 1) modparam("tls", "require_certificate", 1) i restart kamailio: "kamctl restart" log in tail -f /var/log/message Sep 4 05:17:42 appliance /usr/local/sbin/kamailio[3103]: INFO: tls [tls_domain.c:175]: TLSc<default>: tls_method=9 Sep 4 05:17:42 appliance /usr/local/sbin/kamailio[3103]: INFO: tls [tls_domain.c:185]: TLSc<default>: certificate='/usr/local/etc/kamailio//tls/user/user-cert.pem' Sep 4 05:17:42 appliance /usr/local/sbin/kamailio[3103]: INFO: tls [tls_domain.c:190]: TLSc<default>: ca_list='/usr/local/etc/kamailio//tls/user/user-calist.pem' Sep 4 05:17:42 appliance /usr/local/sbin/kamailio[3103]: INFO: tls [tls_domain.c:193]: TLSc<default>: require_certificate=1 Sep 4 05:17:42 appliance /usr/local/sbin/kamailio[3103]: INFO: tls [tls_domain.c:198]: TLSc<default>: cipher_list='(null)' Sep 4 05:17:42 appliance /usr/local/sbin/kamailio[3103]: INFO: tls [tls_domain.c:203]: TLSc<default>: private_key='/usr/local/etc/kamailio//tls/user/user-privkey.pem' Sep 4 05:17:42 appliance /usr/local/sbin/kamailio[3103]: INFO: tls [tls_domain.c:206]: TLSc<default>: verify_certificate=1 Sep 4 05:17:42 appliance /usr/local/sbin/kamailio[3103]: INFO: tls [tls_domain.c:209]: TLSc<default>: verify_depth=9 Sep 4 05:17:42 appliance /usr/local/sbin/kamailio[3103]: INFO: tls [tls_domain.c:331]: TLSc<default>: Server MUST present valid certificate Sep 4 05:17:42 appliance /usr/local/sbin/kamailio[3103]: WARNING: tls [tls_domain.c:395]: tls: set_ssl_options: openssl SSL_OP_TLS_BLOCK_PADDING bug workaround enabled (openssl version 90802f) Sep 4 05:17:42 appliance /usr/local/sbin/kamailio[3116]: INFO: ctl [io_listener.c:224]: io_listen_loop: using epoll_lt io watch method (config) i see that kamailio start okie, but sip phone cannot register. log in :tail -f /var/log/message: Sep 4 05:18:50 appliance /usr/local/sbin/kamailio[3117]: ERROR: tls [tls_server.c:392]: SSL error:error:14094418:SSL routines:SSL3_READ_BYTES:tlsv1 alert unknown ca in portgo : certificate validation failure. please suggest to fix it, thanks. Peter green

2 8

A piece of history - 9 years SER
by Daniel-Constantin Mierla 06 Sep '10

06 Sep '10

Hello, fyi, today are 9 years since first source code commit of SIP Express Router (SER) happened. It started with: git log --pretty=format:"%h%x09%an%x09%ad%x09%s" --reverse | head -3 512dcd9 Andrei Pelinescu-Onciul Mon Sep 3 21:27:11 2001 +0000 Initial revision 888ca09 Andrei Pelinescu-Onciul Tue Sep 4 01:41:39 2001 +0000 parser seems to work e60a972 Andrei Pelinescu-Onciul Tue Sep 4 20:55:41 2001 +0000 First working release Today, 3286 days later, it counts 14 200 commits __only__ in development branch. I digested some statistics and made them available at: http://bit.ly/9J4EKb There are interesting facts about most productive time frames (i.e., hour, day, ...), evolution of default configuration files, command line parameters, ... IPv6 support is there since 2002, but no convenient public network to use it yet. Enjoy and enhance the content as you have interesting aspects of history to add... ... and, VERY important, do not forget to start testing 3.1.0, we are in pre-release phase, there are a lot of new features I am sure you want on your production system: http://sip-router.org/wiki/features/new-in-devel Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

2 2

help to fix error when using tls module.
by peter_green lion 06 Sep '10

06 Sep '10

hi all, I have come back with another problem. I have install kamailio with tls module supported. is there any way to check this module run or not ? thanks for help . Peter Green.

1 0

help for SIP client configuration
by zed li 05 Sep '10

05 Sep '10

Dear sir, How are you? I have installed SER and it runs well. Now I am adding a SIP client to make phone call. I downloaded SERCTL and up, but do not know to use it. Could you tell me how to install and use the software? Or another way is to add user to the tables in MySQL? could you tell me what tables we are using and how to do it? I am sorry to bother you, but I searched the Internet and have got no clear solution. Best regards Zongyin

2 1

failure_route send 500 retry later
by Alejandro Mellado G. 05 Sep '10

05 Sep '10

Hi' I'm trying to forward the call to voicemail on sems when the time of response is out. When the user isn't in location, the forward to voicemail work very fine ( $rc = -1 ). But failure_route, doesn't work and send message "500 Retry Later". I'm using kamailio 3.0.2 and I'm probing with: modparam("tm", "fr_timer", 10) modparam("tm", "fr_inv_timer", 15) ... ... failure_route[FAIL_ONE] { #!ifdef WITH_NAT if (is_method("INVITE") && (isbflagset("6") || isflagset(5))) { unforce_rtp_proxy(); } #!endif if (t_is_canceled()) { exit; } if (t_check_status("486|408")) { revert_uri(); avp_db_load("$ruri","$avp(s:email_address)/$email_scheme"); append_hf("P-App-Name: myvoicemail\r\n"); append_hf("P-App-Param: Email-Address=$avp(s:email_address)\r\n"); rewritehostport("sems_host:5080"); append_branch(); t_relay(); } } Thanks. -- Saluda Atte, Alejandro Mauricio Mellado Gatica Escuela de Ingeniería Informática Universidad Católica de Temuco _________________________________

2 4

Question about avp_db_load
by Raúl Alexis Betancor Santana 03 Sep '10

03 Sep '10

Hi all .. I'm trying to change the way avp_db_load works on my current config to load avp's from DB based on $fU and $fd and not based on uuid, so I have try this ... ... modparam("avpops", "avp_table", "usr_preferences") modparam("avpops", "use_domain", 1) ... # load caller avps avp_db_load("$fu/username", "*"); ... With no success .. because the query that is send to the DB is: select value,attribute,type from usr_preferences where username='{$fU value}' instedad of select value,attribute,type from usr_preferences where username='{$fU value}' and domain='{$fd value}' witch is what I expect when using use_domain=1 Any clue about what I'm doing wrong? Best regards -- Raúl Alexis Betancor Santana Dimensión Virtual

2 2

Unable to load IPv6 locations
by Mikko Lehto 03 Sep '10

03 Sep '10

Hi I am doing simple registration test with IPv6 enabled SIP phone. It seems that Kamailio is unable to load back registrations done via v6 socket after restart. Here is (some) content of location table before starting proxy: --- mysql> select username,contact,socket from location; +----------+-----------------------------------------------+---------------------------------+ | username | contact | socket +----------+-----------------------------------------------+---------------------------------+ | ml6 | sip:ml6@[2a00:17b8::55]:5060;line=z79d766z | udp:2A00:17B8:0:0:0:0:0:67:5060 | | ml4 | sip:ml4@89.18.234.88:5060;line=dp77j5yo | udp:89.18.243.167:5060 | --- Here are the error messages during startup: --- 1(26374) DEBUG: usrloc [udomain.c:415]: loading records - cycle [1] 1(26374) ERROR: <core> [main.c:1116]: ERROR: parse_phostport: too many colons in udp:2A00:17B8:0:0:0:0:0:67:5060 1(26374) ERROR: usrloc [udomain.c:314]: bad socket <udp:2A00:17B8:0:0:0:0:0:67:5060> 1(26374) ERROR: usrloc [udomain.c:430]: sipping record for ml6 in table location --- Here are the listen statements from config: --- listen=udp:[2a00:17b8::67]:5060 listen=udp:89.18.243.167:5060 --- DB is MySQL 5.1, OS is Debian Squeeze. Kamailio version is 3.0.3 and it is from kamailio.org APT repository. At least I just found a typo (sipping), but did I also found a bug? -- Mikko Lehto

2 2

help to fix bug in TLS module.
by peter_green lion 03 Sep '10

03 Sep '10

hi all, I have configure to use TLS module,but i have a problem in step : create key. i follow these step: Creating CA certificate ----------------------- 1. create CA dir mkdir ca cd ca 2. create ca dir structure and files (see ca(1)) mkdir demoCA #default CA name, edit /etc/ss/openssl.cnf mkdir demoCA/private mkdir demoCA/newcerts touch demoCA/index.txt echo 01 >demoCA/serial 2. create CA private key openssl genrsa -out demoCA/private/cakey.pem 2048 chmod 600 demoCA/private/cakey.pem 3. create CA self-signed certificate openssl req -out demoCA/cacert.pem -x509 -new -key demoCA/private/cakey.pem Creating a server/client certificate ------------------------------------ 1. create a certificate request (and its private key in privkey.pem) openssl req -out ser1_cert_req.pem -new -nodes WARNING: the organization name should be the same as in the ca certificate. 2. sign it with the ca certificate openssl ca -in ser1_cert_req.pem -out ser1_cert.pem 3. copy ser1_cert.pem to your ser config. dir Setting ser to use the certificate ---------------------------------- 1. create the ca list file: for each of your ca certificates that you intend to use do: cat cacert.pem >>calist.pem 2. copy your ser certificate, private key and ca list file to your intended machine (preferably in your ser cfg. directory, this is the default place ser searches for) 3. set up ser.cfg to use the certificate if your ser certificate name is different from cert.pem or it is not placed in ser cfg. directory, add to your ser.cfg: modparam("tls", "certificate", "/path/cert_file_name") 4. set up ser to use the private key if your private key is not contained in the certificate (or the certificate name is not the default cert.pem), add to your ser.cfg: modparam("tls", "private_key", "/path/private_key_file") 5. set up ser to use the ca list (optional) add to your ser.cfg: modparam("tls", "ca_list", "/path/ca_list_file") 6. set up tls authentication options: modparam("tls", "verify_certificate", 1) modparam("tls", "require_certificate", 1) (for more information see the module parameters documentation) but in step : 2. sign it with the ca certificate(command :openssl ca -in ser1_cert_req.pem -out ser1_cert.pe ) i have error and it openssl can not make file "ser1_cert.pem " the error is : [root@appliance ca]# openssl ca -in ser1_cert_req.pem -out ser1_cert.pem Using configuration from /etc/pki/tls/openssl.cnf Error opening CA private key ../../CA/private/cakey.pem 15231:error:02001002:system library:fopen:No such file or directory:bss_file.c:352:fopen('../../CA/private/cakey.pem','r') 15231:error:20074002:BIO routines:FILE_CTRL:system lib:bss_file.c:354: unable to load CA private key please help me to fix it, or suggest to check and make it work. thanks all. Peter green

2 2

error when install kamailio with TLS module.
by peter_green lion 03 Sep '10

03 Sep '10

hi all, I try to install kamailio with enable TLS module, but it raise error. when i enable module TLS and run command make all,make install, the error is : #make install Makefile.defs defs skipped make[1]: Entering directory `/usr/src/kamailio-3.0.1/modules/tls' MAIN_NAME=kamailio ./sip-router_cert.sh -d /usr/local/etc/kamailio/ hostname: Unknown host ./sip-router_cert.sh: line 142: which: command not found Could not find openssl command Set OPENSSL environment variable properly (see -h for more info) make[1]: *** [install-tls-cert] Error 1 make[1]: Leaving directory `/usr/src/kamailio-3.0.1/modules/tls' make: *** [install-modules] Error 1 [root@appliance kamailio-3.0.1]# please help me find and fix this problem, thanks for help me. Beter_green.

3 5

Using multiple media proxy pools
by Stagg Shelton 02 Sep '10

02 Sep '10

I have a requirement that I hope can be accomplished. I have to connect to three providers who will be delivering a private IP Layer 2 ethernet endpoint. I need to be able to direct the SIP calls via Kamailio and have the RTP media process through mediaproxy. I also need to try and avoid using separate physical media proxy machines for each SIP provider. Is it possible to run multiple mediaproxy dispatchers bound to separate IP addresses, and run multiple relays with IP addresses corresponding to the various dispatcher subnets. Basically I want to identify pools of mediaproxy servers and choose them based on the provider origination or termination. Alternatively we could use RTP Proxy, though we do not have as much experience with RTP Proxy. Any suggestions that you may have is appreciated. Thanks Stagg

2 1

Avoid resolving names by DNS
by Santiago Soares 02 Sep '10

02 Sep '10

Hello, I'm using kamailio as a simple load balancer. My dispatcher list is something like that: 1 sip:server1:5060 1 sip:server2:5060 2 sip:server1:5060 and /etc/hosts is 10.1.1.1 server1 10.1.1.2 server2 When I want to do some maintenance task in one of the servers, server2, for exemple, I simply change /etc/hosts like that: 10.1.1.1 server1 10.1.1.1 server2 And all the traffic is redirected to server 1. It's a simple yet effective way to remove a server from balancing. The thing is that it works on kamailio 1.5.3, but on kamailio 3.0.3 it doesn't work, since kamailio appears to resolv names only by DNS. I receive this message on startup: Sep 1 12:04:32 install /usr/local/sbin/kamailio[16077]: ERROR: dispatcher [dispatch.c:241]: could not resolve server1 Sep 1 12:04:32 install /usr/local/sbin/kamailio[16077]: ERROR: dispatcher [dispatcher.c:247]: no dispatching list loaded from file Sep 1 12:04:32 install /usr/local/sbin/kamailio[16077]: ERROR: <core> [sr_module.c:874]: init_mod(): Error while initializing module dispatcher Is there any way to avoid this behavior, and force kamailio to use /etc/hosts? Thank you. Santiago Soares Fone: (41) 8488-0537

5 11

Re: [SR-Users] [sr-dev] Table version problems
by marius zbihlei 02 Sep '10

02 Sep '10

Raúl Alexis Betancor Santana wrote: > Hi ... jus trying Kam 3.0.3 ... I get blocked on this error: > > ERROR: <core> [db.c:408]: invalid version 0 for table 0 found, expected 6 > (check table structure and table "version") > ERROR: auth_db [authdb_mod.c:250]: error during table version check. > > It should be checking agains subscriber table version ... but I don't know why > it try to found '0' table ... > > if I do a: > > insert into version (table_name,table_version) values('0','6'); > > It get solved ... but thats only a trick. > > Hello, The check is done once (at fixup time) for the www_authorize and proxy_authorize calls. The table name is the second parameter given to those calls. Be sure to use www_authorize("explample.com", "subscriber"); Cheers Marius

2 4

topoh and SDP endpoints
by Alex Balashov 01 Sep '10

01 Sep '10

The topoh module doesn't touch IP-identifying SDP information, does it? Has there been any interest in implementing that down the road? -- Alex Balashov - Principal Evariste Systems LLC 1170 Peachtree Street 12th Floor, Suite 1200 Atlanta, GA 30309 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/

2 2

help to use force_rtp_proxy([flags [, ip_address]]).
by peter_green lion 01 Sep '10

01 Sep '10

hi all, i am a new user kamailio. i have configure kamailio with RTP proxy, but i have a problem in using : force_rtp_proxy("c","192.168.1.10")because i want to change to force to this ip. my configure is : when server receive "200 OK" it change value in "c= IN IP4 <ip rtp server>" to "c = IN IP4 192.168.1.10" i configure as : kamailio.cfg : #!ifdef WITH_NAT if ((isflagset(5) || isbflagset("6")) && status=~"(183)|(2[0-9][0-9])") { force_rtp_proxy("c","192.168.1.10"); } if (isbflagset("6")) { fix_nated_contact(); } #!endif } when i make call from sip a to sip b, sip b answer .trace as : U 115.78.129.190:54337 -> <server ip>:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP <server ip>;branch=z9hG4bK00c.daa8fe8.0. Via: SIP/2.0/UDP 192.168.1.10:50937;received=115.78.129.190;branch=z9hG4bK-d8754z-1f66a816d651d504-1---d8754z-;rport=63930. Record-Route: <sip:<server ip>;lr;nat=yes>. Contact: <sip:102@192.168.1.10:8576;rinstance=8392ffb3fe461110>. To: <sip:102@<server ip>:5060>;tag=d179a842. From: <sip:101@<server ip>:5060>;tag=1c61b708. Call-ID: ZTQzYjZkYzFjODE1MWFlNjIwNmQ2ZGU5MWUxYmM1NjQ.. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, REGISTER, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces. User-Agent: PortGo v6.0, Build 07282010. Content-Length: 237. . v=0. o=- 30452887 30452887 IN IP4 169.254.202.160. s=http://www.portsip.com. c=IN IP4 169.254.202.160. t=0 0. m=audio 21480 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. a=sendrecv. U <server ip>:5060 -> 115.78.129.190:63930 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.1.10:50937;received=115.78.129.190;branch=z9hG4bK-d8754z-1f66a816d651d504-1---d8754z-;rport=63930. Record-Route: <sip:server ip;lr;nat=yes>. Contact: <sip:102@115.78.129.190:54337;rinstance=8392ffb3fe461110>. To: <sip:102@server ip>:5060>;tag=d179a842. From: <sip:101@<server ip>:5060>;tag=1c61b708. Call-ID: ZTQzYjZkYzFjODE1MWFlNjIwNmQ2ZGU5MWUxYmM1NjQ.. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, REGISTER, SUBSCRIBE, INFO. Content-Type: application/sdp. Supported: replaces. User-Agent: PortGo v6.0, Build 07282010. Content-Length: 237. . v=0. o=- 30452887 30452887 IN IP4 169.254.202.160. s=http://www.portsip.com. c=IN IP4 169.254.202.160. t=0 0. m=audio 21480 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. a=sendrecv. thanks for help me. regards. beter_green

2 1

Cluecon 2010 presentation
by Daniel-Constantin Mierla 01 Sep '10

01 Sep '10

Hello, for those interested, I uploaded my presentation at Cluecon on the web site, link: http://bit.ly/cFDNcX My one and a half day presence in Chicago was surprising good for kamailio and ser - people started to get the real value of the project and catch up with latest features. Moreover, the first day had a lot of presentation with kamailio and ser (see agenda at: http://www.cluecon.com/schedule): - Phillip Zimmermann - did his demo with iptel.org sip service - Mahesh Paolini-Subramanya - a clould based system for telephony - Scott Burkett - back-end platform for web telephony - Robin Rodrigues - several nice examples using upcoming 3.1 addition: embedded http server - Irv Shapiro - ifbyphone platform - programmable telephony applications - I ended the list (the link above), no much left for me to say :-) , but I enjoyed chatting before a lot with folks in the breaks Hopefully these presentations will be soon an event's site. I will try to publish more about each one, they were interesting use cases. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com/

1 0

ratelimit module migration from (k) to generic
by Ovidiu Sas 01 Sep '10

01 Sep '10

Hello all, The (k) ratelimit module is an enhanced version of the original (s) ratelimit module. Are there any objections in moving the ratelimit module from (k) to generic and purging the (s) version? Regards, Ovidiu Sas

2 3

help to get ip from contact header in INVITE sip message.
by peter_green lion 01 Sep '10

01 Sep '10

hi all, please help me to get ip add from Contact header in INVITE message. i see that : ct =<sip:101@192.168.1.10:63027;rinstance=85f97591c218086a> have the ip or domain ("192.168.1.10"). so please suggest to get this ip. thanks so much. beter green

3 4

testing phase for 3.1
by Daniel-Constantin Mierla 31 Aug '10

31 Aug '10

Hello, fyi, we entered in testing phase in order to prepare next major release, numbered 3.1.0. Please help to fill the wiki pages for what is new in this version and migration from 3.0.x: http://sip-router.org/wiki/install/3.0.x-to-3.1.x http://sip-router.org/wiki/features/new-in-devel Testing and feedback are very much appreciated! Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com/

6 23

Kamailio 1.5.4 crash
by gwillingham＠comcast.net 31 Aug '10

31 Aug '10

Kamailio 1.5.4 crashes with the following backtrace. (gdb) bt #0 free_to_params (tb=0x79ffc0) at parser/parse_to.c:79 #1 free_to (tb=0x79ffc0) at parser/parse_to.c:88 #2 0x000000000047fe82 in clean_hdr_field (hf=0x2b32776ec6a0) at parser/hf.c:187 #3 0x00002b327492f525 in clean_msg_clone (type=<value optimized out>, trans=<value optimized out>, req=0x2b32776eb510, rpl=0x79f220, code=<value optimized out>) at sip_msg.h:54 #4 run_trans_callbacks (type=<value optimized out>, trans=<value optimized out>, req=0x2b32776eb510, rpl=0x79f220, code=<value optimized out>) at t_hooks.c:245 #5 0x00002b3274949a1d in relay_reply (t=0x2b32776ed130, p_msg=0x79f220, branch=<value optimized out>, msg_status=200, cancel_bitmap=0x7fff5438ecf8) at t_reply.c:1163 #6 0x00002b327494a123 in reply_received (p_msg=0x79f220) at t_reply.c:1511 #7 0x000000000041f08b in forward_reply (msg=0x79f220) at forward.c:576 #8 0x0000000000445453 in receive_msg ( buf=0x719200 "SIP/2.0 200 OK\r\nVia: SIP/2.0/UDP xxx.xxx.xxx.114;branch=z9hG4bKb415.46c549d2.0\r\nVia: SIP/2.0/UDP xx.xxx.xx.xxx:5060;received=xxx.xxx.xx.xxx;branch=z9hG4bK6cd2ebfb;rport=5060\r\nRecord-Route: <sip:xxx.xxx.xxx.x"..., len=1106, rcv_info=0x7fff5438ede0) at receive.c:212 #9 0x00000000004795ee in udp_rcv_loop () at udp_server.c:449 #10 0x000000000042774e in main_loop (argc=3, argv=0x7fff5438eff8) at main.c:774 #11 main (argc=3, argv=0x7fff5438eff8) at main.c:1321 This is occuring on a centos box, 2.6.18-164.11.1.el5 #1 SMP Wed Jan 20 07:32:21 EST 2010 x86_64 x86_64 x86_64 GNU/Linux The really strange thing is the source of the problem appears to be in clean_hdr_field (hf=0x2b32776ec6a0) at parser/hf.c:187 void clean_hdr_field(struct hdr_field* hf) { if (hf->parsed){ switch(hf->type){ ... case HDR_RPID_T: free_to(hf->parsed); break; ... output from gdb shows hf->parsed to be NULL, however the call to free_to(hf->parsed) is being called, from there bad things happen. Thanks in advance. Gene

4 5

Duplicate INVITE Handling
by Geoffrey Mina 30 Aug '10

30 Aug '10

Hello, I have a question about what the proper way to handle a duplicate presentation of an INVITE is. On occasion I am seeing some packet loss and/or timing issues which are causing some of my end-points to retransmit the INVITE. Here is what I am doing in the most basic sense: ITSP (Bandwidth.com) --> INVITE --> KAMAILIO --> DISPATCHER --> Asterisk (B2BUA) I am seeing Bandwidth.com send an INVITE which I already received. I keep track of all running transactions in a htable which has a key of $ci::$cs::$ft (as per RFC 3261). If I get an invite for something which already has a key in the hashmap, I am currently sending a "482 Loop Detected", but I don't think that is correct as it causes the whole call to tear down instead of letting it continue and assuring Bandwidth.com that I received the initial INVITE and am currently working on it. This is what I am currently doing: ## ## Check to make sure we don't already have an active ## transaction for this call-id, c-seq, and from-tag ## RFC3261 - 8.2.2.2 ## ## We are going to add a key for this unique record if one ## doesn't already exist. The key automatically times out ## after 30 seconds, so we need not worry about cleanup ## if($sht(loop_check=>$ci::$cs::$ft) == null){ xlog("L_INFO","No transaction found, adding to our hashtable\n"); $sht(loop_check=>$ci::$cs::$ft) = 1; }else{ xlog("L_ERR","Loop Detected: $ci::$cs::$ft\n"); sl_send_reply("482","Loop Detected - Duplicate Session Presentation"); exit; } Can I just swallow the second INVITE and do an exit; in my script? Should I do an sl_send_reply(100,"Trying")? Any advice would be greatly appreciated. Thanks,

3 5

Re: [SR-Users] kamailio and asterisk proiblem
by Iñaki Baz Castillo 30 Aug '10

30 Aug '10

Please, replies to the maillist. 2010/8/30 <chuks(a)cybernergies.com>: > I have just that on asterisk, host = kamailio IP and insecure=host,invite. I > don't think this is asterisk related. If you have "insecure=host,invite" then it's wrong. -- Iñaki Baz Castillo <ibc(a)aliax.net>

1 0

load_credentials PVs
by Alex Balashov 30 Aug '10

30 Aug '10

Is there a particular reason why in the 'load_credentials' parameter to the 'auth_db' module, one can only store values in AVPs, not script variables? e.g. this works: modparam("auth_db", "load_credentials", "$avp(s:var1)=column") But not this: modparam("auth_db", "load_credentials", "$var(var1)=column"); it results in an error like this: Aug 28 19:00:54 diminuendo-1 /usr/local/sbin/kamailio[19651]: ERROR: auth_db [aaa_avps.h:113]: malformed or non AVP $var(auth_user_id) AVP definition Aug 28 19:00:54 diminuendo-1 /usr/local/sbin/kamailio[19651]: ERROR: auth_db [aaa_avps.h:179]: parse failed in "$var(auth_user_id)" at pos 0($var(auth_user_id)) Aug 28 19:00:54 diminuendo-1 /usr/local/sbin/kamailio[19651]: ERROR: auth_db [authdb_mod.c:208]: failed to parse credentials Aug 28 19:00:54 diminuendo-1 /usr/local/sbin/kamailio[19651]: ERROR: <core> [sr_module.c:874]: init_mod(): Error while initializing module auth_db -- Alex Balashov - Principal Evariste Systems LLC 1170 Peachtree Street 12th Floor, Suite 1200 Atlanta, GA 30309 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/

3 2

Re: [SR-Users] Kamailio 3.0 cann't access the RTPProxy
by Daniel-Constantin Mierla 30 Aug '10

30 Aug '10

On 8/17/10 7:14 PM, KevinJin wrote: > Hello Daniel, > > What does the nathelper error messages mean? > ========== > hmm, invalid argurment ... try with this line: > > LM_ERR("can't send command to a RTP proxy (%s/%d) [sock %d (%d), vcnt > %d]\n", > strerror(errno), errno, rtpp_socks[node->idx], node->idx, vcnt); > > maybe will give some hints about which value is invalid. > > Here's the log after the change: > 4(14415) ERROR: nathelper [nathelper.c:2457]: can't send command to a > RTP proxy (Invalid argument/22) [sock 7 (0), vcnt 18] can you check the manual page for writev on your system to see what are possible reasons for 'Invalid argument' error (error code 22)? The values in between [...] are just some debug values I wanted to see, not error codes or so. Cheers, Daniel > 4(14415) ERROR: nathelper [nathelper.c:2492]: proxy > <udp:210.13.x.y:7722> does not respond, disable it > 4(14415) ERROR: nathelper [nathelper.c:3144]: no available proxies > > Thanks, > Kevin > ------------------------------------------------------------------------ > From: kevin.jzh(a)hotmail.com > To: miconda(a)gmail.com > Subject: RE: [SR-Users] Kamailio 3.0 cann't access the RTPProxy > Date: Tue, 17 Aug 2010 18:51:02 +0800 > > > Hi Daniel, > > Attached are the two kamailio logs, > unix_socket_log --- run rtp proxy wth -s unix:/tmp/rtpproxy.sock > udp_rtpproxy_log --- run rtp proxy with -s udp:*:7722 > > Please help to check what's wrong with it. > > Thanks, > Kevin > ------------------------------------------------------------------------ > Date: Tue, 17 Aug 2010 11:29:46 +0200 > From: miconda(a)gmail.com > To: kevin.jzh(a)hotmail.com > CC: sr-users(a)lists.sip-router.org > Subject: Re: [SR-Users] Kamailio 3.0 cann't access the RTPProxy > > Hello, > > please send full log at startup, your snippets include just few lines > per process, being mixed from different processes. > > Cheers, > Daniel > > > On 8/14/10 3:54 AM, KevinJin wrote: > > Hello, > > ------------------------------------------------------------------------ > Date: Thu, 12 Aug 2010 22:16:18 +0200 > From: miconda(a)gmail.com <mailto:miconda@gmail.com> > To: kevin.jzh(a)hotmail.com <mailto:kevin.jzh@hotmail.com> > CC: sr-users(a)lists.sip-router.org > <mailto:sr-users@lists.sip-router.org> > Subject: Re: [SR-Users] Kamailio 3.0 cann't access the RTPProxy > > Hello, > > On 8/12/10 8:34 PM, KevinJin wrote: > > Hello, > > ------------------------------------------------------------------------ > Date: Thu, 12 Aug 2010 18:46:19 +0200 > From: miconda(a)gmail.com <mailto:miconda@gmail.com> > To: kevin.jzh(a)hotmail.com <mailto:kevin.jzh@hotmail.com> > CC: sr-users(a)lists.sip-router.org > <mailto:sr-users@lists.sip-router.org> > Subject: Re: [SR-Users] Kamailio 3.0 cann't access the RTPProxy > > Hello, > > On 8/12/10 4:47 PM, KevinJin wrote: > > Hi Daniel, > > What does the log below means? Does it mean nathelper has > issue to send the request to RTP proxy first or nathelper > doesn't receive a response after sending a request to the > rtp proxy? > 0(27429) ERROR: nathelper [nathelper.c:2457]: can't send > command to a RTP proxy > > this error is printed when write to socket fails. Do you have > any firewall running on the system? Is the user under which > kamailio runs allowed to write to sockets? > > > There's no firewall on the system, and I run the kamailio > as root, > root 26310 1 0 02:24:19 ? 0:00 > /usr/local/kamailio-3.0.2/sbin/kamailio -f > /usr/local/kamailio-3.0.2/etc/kamail > > You can edit module_k/nathelper/nathelper.c and replace > the line 2457 with: > > LM_ERR("can't send command to a RTP proxy (%s/%d)\n", > strerror(errno), errno); > > Recompile and reinstall. Hopefully will get more hints > about what happens. > > > Here is the error message after the change: > 2(26312) ERROR: nathelper [nathelper.c:2457]: can't send > command to a RTP proxy(Invalid argument/22) > 2(26312) ERROR: nathelper [nathelper.c:2492]: proxy > <udp:210.13.124.15:7722> does not respond, disable it > 2(26312) ERROR: nathelper [nathelper.c:3144]: no available proxies > what could be the cause? > > hmm, invalid argurment ... try with this line: > > LM_ERR("can't send command to a RTP proxy (%s/%d) [sock %d (%d), > vcnt %d]\n", > strerror(errno), errno, rtpp_socks[node->idx], node->idx, vcnt); > > maybe will give some hints about which value is invalid. > > Here's the log after the change: > 4(14415) ERROR: nathelper [nathelper.c:2457]: can't send command > to a RTP proxy (Invalid argument/22) [sock 7 (0), vcnt 18] > 4(14415) ERROR: nathelper [nathelper.c:2492]: proxy > <udp:210.13.x.y:7722> does not respond, disable it > 4(14415) ERROR: nathelper [nathelper.c:3144]: no available proxies > > > Can you try as well with an unix file socket: > > modparam("nathelper", "rtpproxy_sock", "unix:/tmp/rtpproxy.sock") > > then start rtpproxy with -s unix:/tmp/rtpproxy.sock > > > 4(17530) INFO: nathelper [nathelper.c:2369]: rtp proxy > <unix:/tmp/rtpproxy.sock> found, support for it re-enabled > 3(17529) ERROR: nathelper [nathelper.c:2429]: can't send command > to a RTP proxy > 3(17529) ERROR: nathelper [nathelper.c:2492]: proxy > <unix:/tmp/rtpproxy.sock> does not respond, disable it > 3(17529) ERROR: nathelper [nathelper.c:3144]: no available proxies > > Thanks, > Kevin > > I have no solaris (sparc) to try myself... > > Cheers, > Daniel > > > > Test env: > UA1 (Behind NAT) --------> Kamailio & RTPproxy (Public IP) > --------->UA2 (Public IP) > > Thanks, > Kevin > Cheers, > Daniel > > > > 0(27429) ERROR: nathelper [nathelper.c:2492]: proxy > <udp:210.13.124.15:7722> does not respond, disable it > > There's no problem for the resource(CPU, mem etc.) on the > server, the load is very low. > > Thanks in advance! > ---------- > 0(27429) DEBUG: nathelper [nhelpr_funcs.c:148]: type > <application/sdp> found valid > 0(27429) ERROR: nathelper [nathelper.c:3144]: no available > proxies > 0(27429) ERROR: nathelper [nathelper.c:2627]: no available > proxies > 0(27429) DEBUG: nathelper [nhelpr_funcs.c:148]: type > <application/sdp> found valid > 0(27429) INFO: nathelper [nathelper.c:2369]: rtp proxy > <udp:210.13.124.15:7722> found, support for it re-enabled > 0(27429) DEBUG: nathelper [nathelper.c:3196]: proxy reply: > 42040 210.13.124.14 > 0(27429) DEBUG: nathelper [nhelpr_funcs.c:148]: type > <application/sdp> found valid > 0(27429) ERROR: nathelper [nathelper.c:2457]: can't send > command to a RTP proxy > 0(27429) ERROR: nathelper [nathelper.c:2492]: proxy > <udp:210.13.124.15:7722> does not respond, disable it > 0(27429) ERROR: nathelper [nathelper.c:3144]: no available > proxies > 0(27429) ERROR: nathelper [nathelper.c:2627]: no available > proxies > > Thanks, > Kevin > ------------------------------------------------------------------------ > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org> > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > -- > Daniel-Constantin Mierla > http://www.asipto.com/ > > > -- > Daniel-Constantin Mierla > http://www.asipto.com/ > > > -- > Daniel-Constantin Mierla > http://www.asipto.com/ -- Daniel-Constantin Mierla http://www.asipto.com/

2 2

Question
by Stefan Parreiras 30 Aug '10

30 Aug '10

Hello, My name is Stefan, I am from Brazil I created a Iptel SIP account, but I would like to know if is possible make to call to the phone number...Could you help me with this doubt? Regards. Stefan

2 1

lcr
by Anders 30 Aug '10

30 Aug '10

Hi, I some calls to a specific `sub-region` that I want to limit to a specific trunk. Example. Region 1: prefix 55, trunk 1, priority 10 Region 1: prefix 55, trunk 2, priority 20 Sub-region A: prefix 5512, trunk 1, priority 30 I do not want the calls to prefix 5512 to fail-over to trunk 2 in case of 404, 503 or the like. So I have tried to put priority 30 on sub-region A and a lower priority on Region 1, but it still fails over to trunk 2 - can I prevent this? Thanks! //Anders

4 9

help to configre RTP proxy.
by truong ngoc THANH 27 Aug '10

27 Aug '10

hi all, I hae configure RTP proxy it work ok. please help me to configure that : if sip client is in the same NAT(same LAN network) , they can make call directly (RTP go direct without through RTP proxy). thanks so much. ThanhTruong HTK-INC Co.,Ltd VN www.htk-inc.com http://pbxvn.com/forum/

3 7

help to fix bug in nathelper module
by truong ngoc THANH 27 Aug '10

27 Aug '10

hi all, I have configure nathelper in kamailio, but it does not work as I expect. I using function : force_rpt_proxy("","115.X.X.190") to change ip_add of SDP packet. But I see that value c= <ip add > not change to c=<115.X.X.190>. thanks for help. ThanhTruong HTK-INC Co.,Ltd http://www.htk-inc.com/ http://pbxvn.com/forum/

1 0

LCR prefix based
by inge 26 Aug '10

26 Aug '10

Hi, I'm still disapointed after reading the documentation of LCR module. I use SER 0.9.9 and I'm wondering if it's possible to use LCR based on destination prefix ? I mean that if I call 056XXXXX I want to use GW1. For 06XXXX I want to use GW2, etc. Thank you for your opinion. Regards, Adrien

4 4

Kamailio 1.5 - Dispatcher priority usage
by Geoffrey Mina 26 Aug '10

26 Aug '10

Hello, I have a question about the "priority" in the dispatcher module. I looked in the docu-wiki and there was nothing which actually described how this field is used (exactly), so I can only guess or ask the list. Say I have 3 destinations in a set: set: 1 dest: sip:1.2.3.4:5060 priority: 1 set: 1 dest: sip:1.2.3.5:5060 priority: 1 set: 1 dest: sip:1.2.3.6:5060 priority: 0 I am using ds_select_domain("1","4") to retrieve my list of destinations. In this scenario, would the calls only route to 1.2.3.6 if .4 and .5 were both out of service? Thanks.

3 4

help to configure RTP stream with NAT.
by truong ngoc THANH 26 Aug '10

26 Aug '10

hi all, i have using RTP proxy, and i see that RTP stream is handled by RTP proxy. so how to configure in kamailio or which module make RTP stream direct from sip client to another one ? please suggest if anyone know. thanks. TRUONG NGOC THANH Telecommunications Engineer Tel: 0984 480 646 Y!M: ngoc217thanh

4 8

SIP routing problem when kamailio uses 2 interfaces
by Gnaneshwar Gatla 26 Aug '10

26 Aug '10

Hello All, I'm using Kamailio as a Border controller for my VoIP Research project at my school. The problem I'm facing is Kamailio routes the traffic to the private network where my asterisk machine is listening. The asterisk machine responds to the Kamailio using the public network but not the private. Is this an issue with Asterisk? If so how do I resolve it? client kam_public Kam_pvt Ast_pvt Ast_public | | | | | | | | | | | INVITE | | | | |---------------->| | INVITE | | | | |---------->| | | | | | | | | | | | | | | | | | | | 404 | | | |<----------|-----------|-----------| | | | | | | | | ACK | | | | |---------->| | | | | | | | 404 | | | | |<----------------| | | | | | | | | | ACK | | | | |---------------->| | | | | | | | | | | | | | Regards Gnaneshwar

3 2

Re: [SR-Users] removing stuff
by Daniel-Constantin Mierla 25 Aug '10

25 Aug '10

please cc all the time the mailing list: sr-users(a)lists.sip-router.org On 8/25/10 1:16 PM, Anders wrote: > Thanks Daniel. I found out that the sip_trace was pretty straight > forward to turn off, by running "kamctl fifo sip_trace off". So far, > so good - no more massive data collection on that. Now I need to stop > the radius accounting from collecting data to the 'radacct' table. I > want to continue whatever goes to 'acc' - but I don't know if that > data comes from the radius accounting too(?). So what do I do to stop > collecting to 'radacct' but continue to 'acc'? Accounting to radius is done by setting radius flag. Just remove all parameters to acc module that starts with radius_* Cheers, Daniel > Thanks!! > > --Anders > > On Mon, Aug 23, 2010 at 11:09 AM, Daniel-Constantin Mierla > <miconda(a)gmail.com> wrote: >> Hello, >> >> On 8/23/10 9:32 AM, Anders wrote: >>> Hi, >>> >>> This concerns third party tools...I suspect. >>> >>> First - I run Kamailio 1.5 and use Siremis to calculate the calls >>> (actually just the stored procedure and cron job defined, not the >>> interface). >>> >>> I have earlier been messing around with CDRTools and radius, but I >>> don't use these anymore. However, they seem to still be generating >>> lots of data on this server and overloading MySql - so I really want >>> to get rid of them. Specifically are my radacct and sip_trace tables >>> in MySql getting huge, and keep being populated...and I want to stop >>> this 'madness'. What do I do to stop/remove this? without putting the >>> current installation at risk) >> what do you want to clean? the radius accounting and siptrace from your >> config? >> >> If yes, then just edit kamailio config and remove parameters for radius >> accounting and the siptrace module as well as sip_trace() function and >> siptrace parameter from there. >> >> Cheers, >> Daniel >> >> -- >> Daniel-Constantin Mierla >> http://www.asipto.com/ >> >> -- Daniel-Constantin Mierla http://www.asipto.com/

1 0

help to fix error in carrierroute module.
by truong ngoc THANH 25 Aug '10

25 Aug '10

hi all, please help to correct configure in carrierroute module. i have configure carrierroute in kamailio as bellow. ========= kamailio.cfg === route[1] { # route calls based on hash over callid # choose route domain 0 of the default carrier if(!cr_route("default", "default", "$rU", "$rU", "call_id")){ sl_send_reply("403", "Not allowed"); } else { # In case of failure, re-route the request t_on_failure("1"); # Relay the request to the gateway t_relay(); } } failure_route[1] { # In case of failure, send it to an alternative route: if (t_check_status("408|5[0-9][0-9]")) { #choose route domain 1 of the default carrier if(!cr_route("default", "default", "$rU", "$rU", "call_id")){ } else { xlog("failure 2"); t_on_failure("2"); t_relay(); } } } =========\ in database: mysql> select * from carrierroute; +----+---------+--------+-------------+-------+------+------+-------+--------------+----------------+----------------+-------------+ | id | carrier | domain | scan_prefix | flags | mask | prob | strip | rewrite_host | rewrite_prefix | rewrite_suffix | description | +----+---------+--------+-------------+-------+------+------+-------+--------------+----------------+----------------+-------------+ | 1 | 1 | 1 | 1000 | 0 | 0 | 1 | 0 | 192.168.1.42 | | | FRANCE | +----+---------+--------+-------------+-------+------+------+-------+--------------+----------------+----------------+-------------+ 1 row in set (0.00 sec) mysql> select * from carrier_name; +----+---------+ | id | carrier | +----+---------+ | 1 | default | +----+---------+ mysql> select * from domain_name; +----+---------+ | id | domain | +----+---------+ | 1 | default | +----+---------+ ================ in server 192.168.1.40 : subscriber table: mysql> select * from subscriber; +----+----------+--------------+----------+---------------+----------------------------------+----------------------------------+------+ | id | username | domain | password | email_address | ha1 | ha1b | rpid | +----+----------+--------------+----------+---------------+----------------------------------+----------------------------------+------+ | 1 | 1 | 192.168.1.40 | 1 | | b8ee4a7080092d7fd27357de1799b111 | d1343e4f0259199ed6b06d6338d39903 | NULL | | 2 | 2 | 192.168.1.40 | 2 | | 25f61f90a19afe508629127838c11509 | 7f4ec87f39f48d0d8f8a537cfc34c5c9 | NULL | | 3 | 3 | kamailio.com | 3 | | 63a131bf02b4123073970731a4525470 | 64e49a4f89317c5ee757ba0366ec9307 | NULL | | 4 | 1000 | 192.168.1.40 | 1000 | | 8e9f00da8e49cc8b844b6cee1752b949 | 2bff4783b0ce3412bf33e7400fbbedce | NULL | | 5 | 4 | 192.168.1.40 | 4 | | 037cf4277ce31c104e4b1bc7bbcf8ad4 | cb2321d9ac8bfe749cedc1d66cad5d73 | NULL | | 6 | 1 | kamailio.com | 1 | | 31fb6891a2fcb9b13a952ce3f12359c9 | 84add6cffd100f8ca3c8660c3c9617b5 | NULL | | 7 | 1 | 192.168.1.42 | 1 | | ad57b2abdf288984bc4a897a9e7f2ba9 | 60b9ab690724e5a535df794131e2345f | NULL | | 8 | 1000 | 192.168.1.42 | 1000 | | 0a9fb3a1de135a32b9090798a0dcb240 | bf5a95934a59b480dbf85abe386144b6 | NULL | +----+----------+--------------+----------+---------------+----------------------------------+----------------------------------+------+ I try to make call from 1(a)192.168.1.40 to 1000(a)192.168.1.42 but the call failed. sip client 1(a)192.168.1.40 show that : server failure and sip client 1000(a)192.168.1.42 ringing. ============ sip trace for 192.168.1.40: U 192.168.1.40:5060 -> 192.168.1.42:5060 INVITE sip:1000@192.168.1.42 SIP/2.0. Record-Route: <sip:192.168.1.40;lr=on>. Via: SIP/2.0/UDP 192.168.1.40;branch=z9hG4bKf92a.649f39e3.0. Via: SIP/2.0/UDP 192.168.1.10:50495;branch=z9hG4bK-d8754z-ec188605061deb73-1---d8754z-;rport=50495. Max-Forwards: 69. Contact: <sip:1@192.168.1.10:50495;rinstance=9a88ebaf62e2a935>. To: <sip:1000@192.168.1.40:5060>. From: <sip:1@192.168.1.40:5060>;tag=fd3fe92e. Call-ID: MmNjYTZjMjNmOWZkN2RkZWE3NWUxOWZlNWUyOTY3ZmM.. CSeq: 2 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO. Content-Type: application/sdp. Supported: replaces. User-Agent: 3CXPhone 4.0.10858.0. Content-Length: 279. . v=0. o=3cxVCE 96031755 384028260 IN IP4 192.168.1.10. s=3cxVCE Audio Call. c=IN IP4 192.168.1.10. t=0 0. m=audio 40002 RTP/AVP 0 8 3 101. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:3 GSM/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. a=sendrecv. U 192.168.1.40:5060 -> 192.168.1.10:50495 SIP/2.0 500 I'm terribly sorry, server error occurred (6/SL). Via: SIP/2.0/UDP 192.168.1.10:50495;branch=z9hG4bK-d8754z-ec188605061deb73-1---d8754z-;rport=50495. To: <sip:1000@192.168.1.40:5060>;tag=ae07808f44ebda0182a54cf170e36dbe.ba60. From: <sip:1@192.168.1.40:5060>;tag=fd3fe92e. Call-ID: MmNjYTZjMjNmOWZkN2RkZWE3NWUxOWZlNWUyOTY3ZmM.. CSeq: 2 INVITE. Server: kamailio (3.0.1 (i386/linux)). Content-Length: 0. . U 192.168.1.10:50495 -> 192.168.1.40:5060 ACK sip:1000@192.168.1.40:5060 SIP/2.0. Via: SIP/2.0/UDP 192.168.1.10:50495;branch=z9hG4bK-d8754z-ec188605061deb73-1---d8754z-;rport. Max-Forwards: 70. To: <sip:1000@192.168.1.40:5060>;tag=ae07808f44ebda0182a54cf170e36dbe.ba60. From: <sip:1@192.168.1.40:5060>;tag=fd3fe92e. Call-ID: MmNjYTZjMjNmOWZkN2RkZWE3NWUxOWZlNWUyOTY3ZmM.. CSeq: 2 ACK. Content-Length: 0. and sip trace in 192.168.1.42: ==== U 192.168.1.40:5060 -> 192.168.1.42:5060 INVITE sip:1000@192.168.1.42 SIP/2.0. Record-Route: <sip:192.168.1.40;lr=on>. Via: SIP/2.0/UDP 192.168.1.40;branch=z9hG4bKc4e.5ee44227.0. Via: SIP/2.0/UDP 192.168.1.10:50495;branch=z9hG4bK-d8754z-7e347d55637cf134-1---d8754z-;rport=50495. Max-Forwards: 69. Contact: <sip:1@192.168.1.10:50495;rinstance=9a88ebaf62e2a935>. To: <sip:1000@192.168.1.40:5060>. From: <sip:1@192.168.1.40:5060>;tag=722dc40c. Call-ID: ZGY0YTY5YmYzM2E2MzEzNDgzODhkMTllN2RiY2ExOGM.. CSeq: 2 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO. Content-Type: application/sdp. Supported: replaces. User-Agent: 3CXPhone 4.0.10858.0. Content-Length: 280. . v=0. o=3cxVCE 227296140 367695825 IN IP4 192.168.1.10. s=3cxVCE Audio Call. c=IN IP4 192.168.1.10. t=0 0. m=audio 40000 RTP/AVP 0 8 3 101. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:3 GSM/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. a=sendrecv. U 192.168.1.42:5060 -> 192.168.1.40:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 192.168.1.40;branch=z9hG4bKc4e.5ee44227.0. Via: SIP/2.0/UDP 192.168.1.10:50495;branch=z9hG4bK-d8754z-7e347d55637cf134-1---d8754z-;rport=50495. To: <sip:1000@192.168.1.40:5060>. From: <sip:1@192.168.1.40:5060>;tag=722dc40c. Call-ID: ZGY0YTY5YmYzM2E2MzEzNDgzODhkMTllN2RiY2ExOGM.. CSeq: 2 INVITE. Server: kamailio (3.0.1 (i386/linux)). Content-Length: 0. . U 192.168.1.42:5060 -> 192.168.1.10:6872 INVITE sip:1000@192.168.1.10:6872;rinstance=c308344d9524d1a4 SIP/2.0. Record-Route: <sip:192.168.1.42;lr=on>. Record-Route: <sip:192.168.1.40;lr=on>. Via: SIP/2.0/UDP 192.168.1.42;branch=z9hG4bKc4e.f9fad397.0. Via: SIP/2.0/UDP 192.168.1.40;branch=z9hG4bKc4e.5ee44227.0. Via: SIP/2.0/UDP 192.168.1.10:50495;branch=z9hG4bK-d8754z-7e347d55637cf134-1---d8754z-;rport=50495. Max-Forwards: 68. Contact: <sip:1@192.168.1.10:50495;rinstance=9a88ebaf62e2a935>. To: <sip:1000@192.168.1.40:5060>. From: <sip:1@192.168.1.40:5060>;tag=722dc40c. Call-ID: ZGY0YTY5YmYzM2E2MzEzNDgzODhkMTllN2RiY2ExOGM.. CSeq: 2 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO. Content-Type: application/sdp. Supported: replaces. User-Agent: 3CXPhone 4.0.10858.0. Content-Length: 280. . v=0. o=3cxVCE 227296140 367695825 IN IP4 192.168.1.10. s=3cxVCE Audio Call. c=IN IP4 192.168.1.10. t=0 0. m=audio 40000 RTP/AVP 0 8 3 101. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:3 GSM/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. a=sendrecv. U 192.168.1.10:6872 -> 192.168.1.42:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 192.168.1.42;branch=z9hG4bKc4e.f9fad397.0. Via: SIP/2.0/UDP 192.168.1.40;branch=z9hG4bKc4e.5ee44227.0. Via: SIP/2.0/UDP 192.168.1.10:50495;branch=z9hG4bK-d8754z-7e347d55637cf134-1---d8754z-;rport=50495. Record-Route: <sip:192.168.1.42;lr>. Record-Route: <sip:192.168.1.40;lr=on>. Contact: <sip:1000@192.168.1.10:6872;rinstance=c308344d9524d1a4>. To: <sip:1000@192.168.1.40:5060>;tag=660be63a. From: <sip:1@192.168.1.40:5060>;tag=722dc40c. Call-ID: ZGY0YTY5YmYzM2E2MzEzNDgzODhkMTllN2RiY2ExOGM.. CSeq: 2 INVITE. User-Agent: PortGo v6.0, Build 07282010. Content-Length: 0. ======================== and in server 192.168.1.42:subscriber table: mysql> select * from subscriber; +----+----------+--------------+----------+---------------+----------------------------------+----------------------------------+------+ | id | username | domain | password | email_address | ha1 | ha1b | rpid | +----+----------+--------------+----------+---------------+----------------------------------+----------------------------------+------+ | 1 | 1000 | 192.168.1.42 | 1000 | | 0a9fb3a1de135a32b9090798a0dcb240 | bf5a95934a59b480dbf85abe386144b6 | NULL | | 2 | 1 | 192.168.1.42 | 1 | | ad57b2abdf288984bc4a897a9e7f2ba9 | 60b9ab690724e5a535df794131e2345f | NULL | | 3 | 1 | 192.168.1.40 | 1 | | b8ee4a7080092d7fd27357de1799b111 | d1343e4f0259199ed6b06d6338d39903 | NULL | please suggest to get correct configue. thanks for looking. TRUONG NGOC THANH Telecommunications Engineer Tel: 0984 480 646begin_of_the_skype_highlighting 0984 480 646 end_of_the_skype_highlighting Y!M: ngoc217thanh

2 2

Web and SIP Telephony
by Doddle WebPhone 24 Aug '10

24 Aug '10

This may be useful for calling Kamailio SIP server right from web pages / browsers ( webphone/click2call) For Instance, a web phone link to call: <a href=" http://widget.doddlephone.com/embed/web3phone.jsp?sipserver=proxy.ideasip.c…<http://widget.doddlephone.com/embed/web3phone.jsp?sipserver=proxy.ideasip.c…> "> Tel: +1 234 567 890 </a> We can embed it as a widget/iframe or even deploying/hosting the web-driven phone on our Web servers and integrating with our applications ( for instance, filling such SIP parameters dynamically with PHP, .NET, Ruby,etc) Sergio

2 4

removing stuff
by Anders 23 Aug '10

23 Aug '10

Hi, This concerns third party tools...I suspect. First - I run Kamailio 1.5 and use Siremis to calculate the calls (actually just the stored procedure and cron job defined, not the interface). I have earlier been messing around with CDRTools and radius, but I don't use these anymore. However, they seem to still be generating lots of data on this server and overloading MySql - so I really want to get rid of them. Specifically are my radacct and sip_trace tables in MySql getting huge, and keep being populated...and I want to stop this 'madness'. What do I do to stop/remove this? without putting the current installation at risk) Thanks!! //Anders

2 1

Re: [SR-Users] Kamailio-3.0.3 Config file error‏
by Anthony Mark 23 Aug '10

23 Aug '10

Dear Sir, I just wanted to inform the good people on this forum that i had corrected the problem with the Config file, and the server is running. I noticed on going over the file again that i Fat fingered some keys ;) Sorry for any inconvenience my post may have caused anyone. Thanks for the good work you are doing. Best Regards, Anthony.

1 0

Kamailio-3.0.3 Config file error
by Anthony Mark 23 Aug '10

23 Aug '10

Hi, I made an attempt at installing kamailio-3.0.3 in a box yesterday and ran into a config file error on testing the installation. 0(3232): <core> [cfg.y:3333] parse error in config file /usr/local/etc/kamailio/kamailio.cfg line 24, column 6 : synthax error 0(3232): <core> [cfg.y:3333] parse error in config file /usr/local/etc/kamailio/kamailio.cfg line 74, column 6 : unkown config variable 0(3232): <core> [cfg.y:3333] parse error in config file /usr/local/etc/kamailio/kamailio.cfg line 74, column 7 : For me, this error is unexplainable as i am yet to find anything wrong with the lines in question. ( besides, i did not even touch the lines it says are in error ) But for sake of clearity, please bear with me as i post the Much of the config file as i would like to be corrected and assisted. Thanks in anticipation Anthony. - - - - - - - - - - - - - - - - - - #!KAMAILIO # #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB # # $Id$ # # Kamailio (OpenSER) SIP Server v3.0 - basic configuration script # - web: http://www.kamailio.org # - git: http://sip-router.org # # Direct your questions about this file to: <users(a)lists.kamailio.org> # # Refer to the Core CookBook at http://www.kamailio.org/dokuwiki/doku.php # for an explanation of possible statements, functions and parameters. # # Several features can be enabled using ' #!define WITH_FEATURE' directives: # # *** To run in debug mode: # - define WITH_DEBUG # # *** To enable mysql: - define WITH_MYSQL # # *** To enable authentication execute: - enable mysql - define WITH_AUTH - add users using 'kamctl' # # *** To enable persistent user location execute: - enable mysql - define WITH_USRLOCDB # # *** To enable presence server execute: # - enable mysql # - define WITH_PRESENCE # # *** To enable nat traversal execute: # - define WITH_NAT # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # # *** To enable PSTN gateway routing execute: # - define WITH_PSTN # - set the value of pstn.gw_ip # - check route[PSTN] for regexp routing condition # # *** To enhance accounting execute: - enable mysql - define WITH_ACCDB - add following columns to database !ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; !endif ####### Global Parameters ######### #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!elsedebug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=4 /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to disable the auto discovery of local aliases based on revers DNS on IPs (default on) */#auto_aliases=no port=5060 /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ #listen=udp:10.0.0.10:5060 ####### Custom Parameters ######### # These parameters can be modified runtime via RPC interface # - see the documentation of 'cfg_rpc' module. # # Format: group.id = value 'desc' description # Access: $sel(cfg_get.group.id) or @cfg_get.group.id # #!ifdef WITH_PSTN # PSTN GW Routing # # - pstn.gw_ip: valid IP or hostname as string value, example: # pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" # # - by default is empty to avoid misrouting pstn.gw_ip = "" desc "PSTN GW Address" #!endif ####### Modules Section ######## #set module path mpath="/usr/local/lib/kamailio/modules_k/:/usr/local/lib/kamailio/modules/" /* uncomment next line for MySQL DB support */ !ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "mi_fifo.so" loadmodule "kex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "uri_db.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "mi_rpc.so" loadmodule "acc.so" #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!endif /* uncomment next line for aliases support NOTE: a DB (like db_mysql) module must be also loaded */ #loadmodule "alias_db.so" /* uncomment next line for multi-domain support NOTE: a DB (like db_mysql) module must be also loaded NOTE: be sure and enable multi-domain support in all used modules (see "multi-module params" section ) */ #loadmodule "domain.so" #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif

1 0

opensips long shutdown time
by Dave Singer 22 Aug '10

22 Aug '10

sometimes when I'm do a restart on opensips (using init.d script, with some customization to handle this problem) opensips takes quite a bit of time, like 30 - 50 seconds to stop. Other times it is very quick. I'm using 1.6.2 and 1.6.3. The servers are fairly busy, less then 200 calls per sec, with around 1000 sustained open calls. I am using dialog module and db mode is 3 (on shutdown) with postgresql 8.4. However since sometimes it is very quick with the same call volume it doesn't "feel" like this is the problem. Further I have had it be slow to stop when testing a config that is not using dialog at all and there are no active calls or any sip activity when I try to stop it. My hunch is slightly toward it is waiting on a transaction but not so sure. The pertinent part of my init.d script customizations are for after sending opensips the kill <pid> signal to every 0.1 seconds check if it has stopped yet then start it back up. Needless to say, having it take 50 seconds to restart, and not responding to sip traffic does not make customers happy when your dealing with thousands of call. :( Any thoughts or things to try would be appreciated. Thanks cando

2 1

help to error when configure carrierroute in kamailio 3.0.1
by truong ngoc THANH 20 Aug '10

20 Aug '10

hi all, i have configure carrierroute in kamailio as bellow. ========= kamailio.cfg === route[1] { # route calls based on hash over callid # choose route domain 0 of the default carrier if(!cr_route("default", "default", "$rU", "$rU", "call_id")){ sl_send_reply("403", "Not allowed"); } else { # In case of failure, re-route the request t_on_failure("1"); # Relay the request to the gateway t_relay(); } } failure_route[1] { # In case of failure, send it to an alternative route: if (t_check_status("408|5[0-9][0-9]")) { #choose route domain 1 of the default carrier if(!cr_route("default", "default", "$rU", "$rU", "call_id")){ } else { xlog("failure 2"); t_on_failure("2"); t_relay(); } } } =========\ in database: mysql> select * from carrierroute; +----+---------+--------+-------------+-------+------+------+-------+--------------+----------------+----------------+-------------+ | id | carrier | domain | scan_prefix | flags | mask | prob | strip | rewrite_host | rewrite_prefix | rewrite_suffix | description | +----+---------+--------+-------------+-------+------+------+-------+--------------+----------------+----------------+-------------+ | 1 | 1 | 1 | 1000 | 0 | 0 | 1 | 0 | 192.168.1.42 | | | FRANCE | +----+---------+--------+-------------+-------+------+------+-------+--------------+----------------+----------------+-------------+ 1 row in set (0.00 sec) mysql> select * from carrier_name; +----+---------+ | id | carrier | +----+---------+ | 1 | default | +----+---------+ mysql> select * from domain_name; +----+---------+ | id | domain | +----+---------+ | 1 | default | +----+---------+ ================ in server 192.168.1.40 : subscriber table: mysql> select * from subscriber; +----+----------+--------------+----------+---------------+----------------------------------+----------------------------------+------+ | id | username | domain | password | email_address | ha1 | ha1b | rpid | +----+----------+--------------+----------+---------------+----------------------------------+----------------------------------+------+ | 1 | 1 | 192.168.1.40 | 1 | | b8ee4a7080092d7fd27357de1799b111 | d1343e4f0259199ed6b06d6338d39903 | NULL | | 2 | 2 | 192.168.1.40 | 2 | | 25f61f90a19afe508629127838c11509 | 7f4ec87f39f48d0d8f8a537cfc34c5c9 | NULL | | 3 | 3 | kamailio.com | 3 | | 63a131bf02b4123073970731a4525470 | 64e49a4f89317c5ee757ba0366ec9307 | NULL | | 4 | 1000 | 192.168.1.40 | 1000 | | 8e9f00da8e49cc8b844b6cee1752b949 | 2bff4783b0ce3412bf33e7400fbbedce | NULL | | 5 | 4 | 192.168.1.40 | 4 | | 037cf4277ce31c104e4b1bc7bbcf8ad4 | cb2321d9ac8bfe749cedc1d66cad5d73 | NULL | | 6 | 1 | kamailio.com | 1 | | 31fb6891a2fcb9b13a952ce3f12359c9 | 84add6cffd100f8ca3c8660c3c9617b5 | NULL | | 7 | 1 | 192.168.1.42 | 1 | | ad57b2abdf288984bc4a897a9e7f2ba9 | 60b9ab690724e5a535df794131e2345f | NULL | | 8 | 1000 | 192.168.1.42 | 1000 | | 0a9fb3a1de135a32b9090798a0dcb240 | bf5a95934a59b480dbf85abe386144b6 | NULL | +----+----------+--------------+----------+---------------+----------------------------------+----------------------------------+------+ I try to make call from 1(a)192.168.1.40 to 1000(a)192.168.1.42 but the call failed. sip client 1(a)192.168.1.40 show that : server failure and sip client 1000(a)192.168.1.42 ringing. ============ sip trace for 192.168.1.40: U 192.168.1.40:5060 -> 192.168.1.42:5060 INVITE sip:1000@192.168.1.42 SIP/2.0. Record-Route: <sip:192.168.1.40;lr=on>. Via: SIP/2.0/UDP 192.168.1.40;branch=z9hG4bKf92a.649f39e3.0. Via: SIP/2.0/UDP 192.168.1.10:50495;branch=z9hG4bK-d8754z-ec188605061deb73-1---d8754z-;rport=50495. Max-Forwards: 69. Contact: <sip:1@192.168.1.10:50495;rinstance=9a88ebaf62e2a935>. To: <sip:1000@192.168.1.40:5060>. From: <sip:1@192.168.1.40:5060>;tag=fd3fe92e. Call-ID: MmNjYTZjMjNmOWZkN2RkZWE3NWUxOWZlNWUyOTY3ZmM.. CSeq: 2 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO. Content-Type: application/sdp. Supported: replaces. User-Agent: 3CXPhone 4.0.10858.0. Content-Length: 279. . v=0. o=3cxVCE 96031755 384028260 IN IP4 192.168.1.10. s=3cxVCE Audio Call. c=IN IP4 192.168.1.10. t=0 0. m=audio 40002 RTP/AVP 0 8 3 101. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:3 GSM/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. a=sendrecv. U 192.168.1.40:5060 -> 192.168.1.10:50495 SIP/2.0 500 I'm terribly sorry, server error occurred (6/SL). Via: SIP/2.0/UDP 192.168.1.10:50495;branch=z9hG4bK-d8754z-ec188605061deb73-1---d8754z-;rport=50495. To: <sip:1000@192.168.1.40:5060>;tag=ae07808f44ebda0182a54cf170e36dbe.ba60. From: <sip:1@192.168.1.40:5060>;tag=fd3fe92e. Call-ID: MmNjYTZjMjNmOWZkN2RkZWE3NWUxOWZlNWUyOTY3ZmM.. CSeq: 2 INVITE. Server: kamailio (3.0.1 (i386/linux)). Content-Length: 0. . U 192.168.1.10:50495 -> 192.168.1.40:5060 ACK sip:1000@192.168.1.40:5060 SIP/2.0. Via: SIP/2.0/UDP 192.168.1.10:50495;branch=z9hG4bK-d8754z-ec188605061deb73-1---d8754z-;rport. Max-Forwards: 70. To: <sip:1000@192.168.1.40:5060>;tag=ae07808f44ebda0182a54cf170e36dbe.ba60. From: <sip:1@192.168.1.40:5060>;tag=fd3fe92e. Call-ID: MmNjYTZjMjNmOWZkN2RkZWE3NWUxOWZlNWUyOTY3ZmM.. CSeq: 2 ACK. Content-Length: 0. and sip trace in 192.168.1.42: ==== U 192.168.1.40:5060 -> 192.168.1.42:5060 INVITE sip:1000@192.168.1.42 SIP/2.0. Record-Route: <sip:192.168.1.40;lr=on>. Via: SIP/2.0/UDP 192.168.1.40;branch=z9hG4bKc4e.5ee44227.0. Via: SIP/2.0/UDP 192.168.1.10:50495;branch=z9hG4bK-d8754z-7e347d55637cf134-1---d8754z-;rport=50495. Max-Forwards: 69. Contact: <sip:1@192.168.1.10:50495;rinstance=9a88ebaf62e2a935>. To: <sip:1000@192.168.1.40:5060>. From: <sip:1@192.168.1.40:5060>;tag=722dc40c. Call-ID: ZGY0YTY5YmYzM2E2MzEzNDgzODhkMTllN2RiY2ExOGM.. CSeq: 2 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO. Content-Type: application/sdp. Supported: replaces. User-Agent: 3CXPhone 4.0.10858.0. Content-Length: 280. . v=0. o=3cxVCE 227296140 367695825 IN IP4 192.168.1.10. s=3cxVCE Audio Call. c=IN IP4 192.168.1.10. t=0 0. m=audio 40000 RTP/AVP 0 8 3 101. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:3 GSM/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. a=sendrecv. U 192.168.1.42:5060 -> 192.168.1.40:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 192.168.1.40;branch=z9hG4bKc4e.5ee44227.0. Via: SIP/2.0/UDP 192.168.1.10:50495;branch=z9hG4bK-d8754z-7e347d55637cf134-1---d8754z-;rport=50495. To: <sip:1000@192.168.1.40:5060>. From: <sip:1@192.168.1.40:5060>;tag=722dc40c. Call-ID: ZGY0YTY5YmYzM2E2MzEzNDgzODhkMTllN2RiY2ExOGM.. CSeq: 2 INVITE. Server: kamailio (3.0.1 (i386/linux)). Content-Length: 0. . U 192.168.1.42:5060 -> 192.168.1.10:6872 INVITE sip:1000@192.168.1.10:6872;rinstance=c308344d9524d1a4 SIP/2.0. Record-Route: <sip:192.168.1.42;lr=on>. Record-Route: <sip:192.168.1.40;lr=on>. Via: SIP/2.0/UDP 192.168.1.42;branch=z9hG4bKc4e.f9fad397.0. Via: SIP/2.0/UDP 192.168.1.40;branch=z9hG4bKc4e.5ee44227.0. Via: SIP/2.0/UDP 192.168.1.10:50495;branch=z9hG4bK-d8754z-7e347d55637cf134-1---d8754z-;rport=50495. Max-Forwards: 68. Contact: <sip:1@192.168.1.10:50495;rinstance=9a88ebaf62e2a935>. To: <sip:1000@192.168.1.40:5060>. From: <sip:1@192.168.1.40:5060>;tag=722dc40c. Call-ID: ZGY0YTY5YmYzM2E2MzEzNDgzODhkMTllN2RiY2ExOGM.. CSeq: 2 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO. Content-Type: application/sdp. Supported: replaces. User-Agent: 3CXPhone 4.0.10858.0. Content-Length: 280. . v=0. o=3cxVCE 227296140 367695825 IN IP4 192.168.1.10. s=3cxVCE Audio Call. c=IN IP4 192.168.1.10. t=0 0. m=audio 40000 RTP/AVP 0 8 3 101. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:3 GSM/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. a=sendrecv. U 192.168.1.10:6872 -> 192.168.1.42:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 192.168.1.42;branch=z9hG4bKc4e.f9fad397.0. Via: SIP/2.0/UDP 192.168.1.40;branch=z9hG4bKc4e.5ee44227.0. Via: SIP/2.0/UDP 192.168.1.10:50495;branch=z9hG4bK-d8754z-7e347d55637cf134-1---d8754z-;rport=50495. Record-Route: <sip:192.168.1.42;lr>. Record-Route: <sip:192.168.1.40;lr=on>. Contact: <sip:1000@192.168.1.10:6872;rinstance=c308344d9524d1a4>. To: <sip:1000@192.168.1.40:5060>;tag=660be63a. From: <sip:1@192.168.1.40:5060>;tag=722dc40c. Call-ID: ZGY0YTY5YmYzM2E2MzEzNDgzODhkMTllN2RiY2ExOGM.. CSeq: 2 INVITE. User-Agent: PortGo v6.0, Build 07282010. Content-Length: 0. ======================== and in server 192.168.1.42:subscriber table: mysql> select * from subscriber; +----+----------+--------------+----------+---------------+----------------------------------+----------------------------------+------+ | id | username | domain | password | email_address | ha1 | ha1b | rpid | +----+----------+--------------+----------+---------------+----------------------------------+----------------------------------+------+ | 1 | 1000 | 192.168.1.42 | 1000 | | 0a9fb3a1de135a32b9090798a0dcb240 | bf5a95934a59b480dbf85abe386144b6 | NULL | | 2 | 1 | 192.168.1.42 | 1 | | ad57b2abdf288984bc4a897a9e7f2ba9 | 60b9ab690724e5a535df794131e2345f | NULL | | 3 | 1 | 192.168.1.40 | 1 | | b8ee4a7080092d7fd27357de1799b111 | d1343e4f0259199ed6b06d6338d39903 | NULL | please suggest to get correct configue. thanks for looking. TRUONG NGOC THANH Telecommunications Engineer Tel: 0984 480 646 Y!M: ngoc217thanh

1 0

Kamailio v3.0.3 Released
by Daniel-Constantin Mierla 19 Aug '10

19 Aug '10

Kamailio v3.0.3 is out – this is a minor release including fixes in code and documentation since v3.0.2 (configuration file and database compatibility is preserved). Kamailio (OpenSER) 3.0.3 is based on the latest version of GIT branch 3.0, therefore those running 3.0.0, 3.0.1 or 3.0.2 are advised to upgrade — there is no change required to be done to configuration file or database. Source tarballs are available at: http://www.kamailio.org/pub/kamailio/3.0.3/src/ Detailed changelog: http://www.kamailio.org/pub/kamailio/3.0.3/ChangeLog Download via GIT: # git clone –depth 1 git://git.sip-router.org/sip-router kamailio # cd kamailio # git checkout -b kamailio_3.0 origin/kamailio_3.0 Binaries and packages will be uploaded at: http://www.kamailio.org/pub/kamailio/3.0.3/ Modules’ documentation: http://www.kamailio.org/docs/modules/3.0.x/ What is new in 3.0.x release series is summarized in the announcement of v3.0.0: http://www.kamailio.org/w/kamailio-openser-v3.0.0-release-notes/ If you are eager to see what’s up to next major release (v3.1.0, scheduled for September 2010, now being the testing phase), visit: http://sip-router.org/wiki/features/new-in-devel Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com/

1 0

Re: [SR-Users] help to config cfg_db module in kamailio 3.0.1
by truong ngoc THANH 19 Aug '10

19 Aug '10

dear Daniel, when i add xlog("my var value is: $sel(cfg_get.my.route_addr)\n"); i see that kamailio load right value. my scenarios as: my kamailio server(192.168.1.40) and the $sel(cfg_get.my.route_addr) is 192.168.1.110 i register with 1(a)192.168.1.40 and 2(a)192.168.1.40 then i make call is ok so the route ip is ??? and why i can make call ? TRUONG NGOC THANH Telecommunications Engineer Tel: 0984 480 646 Y!M: ngoc217thanh ________________________________

2 2

preparing kamailio 3.0.3
by Daniel-Constantin Mierla 19 Aug '10

19 Aug '10

Hello, I am planning to release minor version 3.0.3 by beginning of next week. Please commit there what you think it needs to be fixed for this release. Thanks, Daniel -- Daniel-Constantin Mierla http://www.asipto.com/

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Re: [SR-Users] help to config cfg_db module in kamailio 3.0.1
by Daniel-Constantin Mierla 18 Aug '10

18 Aug '10

Hello, first, keep the mailing list on cc, otherwise future emails will be ignored. Then, the -ddd and -E parameters are for kamailio, not kamctl. Just run: kamailio -ddd -E of course, if you have a custom path to your config, keep you -c /path/to/cfg parameter. Cheers, Daniel On 8/18/10 6:20 PM, truong ngoc THANH wrote: > thanks Daniel, > thanks for help, but i cannot start :kamctl -ddd -E > or kamctl -ddd > or kamctl -E. > and can you show me how it work, or how the result when i declare the > parameter, and how do I know the parameter is used for ? > so nice too get your help. > TRUONG NGOC THANH > Telecommunications Engineer > Tel: 0984 480 646 > Y!M: ngoc217thanh > > > ------------------------------------------------------------------------ > *From:* Daniel-Constantin Mierla <miconda(a)gmail.com> > *To:* truong ngoc THANH <ngoc217thanh(a)yahoo.com> > *Cc:* kamailio <sr-users(a)lists.sip-router.org> > *Sent:* Wed, August 18, 2010 10:47:58 PM > *Subject:* Re: [SR-Users] help to config cfg_db module in kamailio 3.0.1 > > Hello, > > not a user of this module, however, start kamailio with parameters > -ddd -E and paste the output here. > > Do you declare my.route_addr in config file as well? Might be that the > parameter is declared in cfg_custom table, the startup debug log > should give more hints. > > Thanks, > Daniel > > > On 8/18/10 7:13 AM, truong ngoc THANH wrote: >> >> Hi all , >> >> I try to configure with cfg_db, but after create and insert value >> into tables in mysql , I cannot start kamailio. >> >> I add My config in kamalio.cfg >> >> ============== >> >> Loadmodule “cfg_db.so <http://cfg_db.so>” >> >> modparam("cfg_db", "db_url", >> "mysql://openser:openserrw@localhost/kamailio") >> >> modparam("cfg_db", "transl_tbl", "cfg_transl") >> >> modparam("cfg_db", "custom_tbl", "cfg_custom") >> >> I add and insert in mysql as: >> >> =============== >> >> cfg_transl table: >> >> group_name|name|cfg_table|cfg_table_group_name_field|cfg_table_name_field|cfg_table_value_field >> >> core|use_dst_blacklist|cfg_dns||| >> >> core|dst_blacklist_mem|cfg_dns||| >> >> core|dst_blacklist_expire||| >> >> my|route_addr|cfg_my|my_group_name|my_name|my_value ; overrides >> <default> values >> >> my|use_rtp_proxy|cfg_my|my_group_name|my_name|my_value >> >> *|*|||| ; matches all param ids, points to tables defined in row >> <default> >> >> <default>||cfg_var|group_name|name|value ; default cfg_table* values >> >> cfg_custom table: >> >> group_name|name|value_type|min_value|max_value|description >> >> my|route_addr|S|||Routing address >> >> my|use_rtp_proxy|I|0|1|Use RTP proxy >> >> cfg_var table: >> >> group_name|my_name|value >> >> core|debug|4 >> >> cfg_my table: >> >> my_group_name|my_name|my_value >> >> my|route_addr|127.0.0.1 >> >> my|use_rtp_proxy|1 >> >> cfg_dns table: >> >> group_name|name|value >> >> dns|use_dst_blacklist|1 >> >> dns|dst_blacklist_mem|1024 >> >> dns|dst_blacklist_expire|60 >> >> when I start kamailio the log show : >> >> =============== >> >> Aug 18 12:04:56 appliance /usr/local/sbin/kamailio[2771]: INFO: >> usrloc [hslot.c:53]: locks array size 512 >> >> Aug 18 12:04:56 appliance /usr/local/sbin/kamailio[2771]: ERROR: >> <core> [cfg/cfg_script.c:80]: ERROR: new_cfg_script_var(): variable >> already exists: my.route_addr >> >> Aug 18 12:04:56 appliance /usr/local/sbin/kamailio[2771]: ERROR: >> cfg_db [cfg_db.c:335]: cfg_db: Error declaring cfg int 'my.route_addr' >> >> Aug 18 12:04:56 appliance /usr/local/sbin/kamailio[2771]: ERROR: >> <core> [sr_module.c:860]: init_mod(): Error while initializing module >> cfg_db >> >> Thanks for looking. >> >> TRUONG NGOC THANH >> Telecommunications Engineer >> Tel: 0984 480 646 >> Y!M: ngoc217thanh >> >> >> >> _______________________________________________ >> 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/

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help to config cfg_db module in kamailio 3.0.1
by truong ngoc THANH 18 Aug '10

18 Aug '10

Hi all , I try to configure with cfg_db, but after create and insert value into tables in mysql , I cannot start kamailio. I add My config in kamalio.cfg ============== Loadmodule “cfg_db.so” modparam("cfg_db", "db_url", "mysql://openser:openserrw@localhost/kamailio") modparam("cfg_db", "transl_tbl", "cfg_transl") modparam("cfg_db", "custom_tbl", "cfg_custom") I add and insert in mysql as: =============== cfg_transl table: group_name|name|cfg_table|cfg_table_group_name_field|cfg_table_name_field|cfg_table_value_field core|use_dst_blacklist|cfg_dns||| core|dst_blacklist_mem|cfg_dns||| core|dst_blacklist_expire||| my|route_addr|cfg_my|my_group_name|my_name|my_value ; overrides <default> values my|use_rtp_proxy|cfg_my|my_group_name|my_name|my_value *|*||||; matches all param ids, points to tables defined in row <default> <default>||cfg_var|group_name|name|value ; default cfg_table* values cfg_custom table: group_name|name|value_type|min_value|max_value|description my|route_addr|S|||Routing address my|use_rtp_proxy|I|0|1|Use RTP proxy cfg_var table: group_name|my_name|value core|debug|4 cfg_my table: my_group_name|my_name|my_value my|route_addr|127.0.0.1 my|use_rtp_proxy|1 cfg_dns table: group_name|name|value dns|use_dst_blacklist|1 dns|dst_blacklist_mem|1024 dns|dst_blacklist_expire|60 when I start kamailio the log show : =============== Aug 18 12:04:56 appliance /usr/local/sbin/kamailio[2771]: INFO: usrloc [hslot.c:53]: locks array size 512 Aug 18 12:04:56 appliance /usr/local/sbin/kamailio[2771]: ERROR: <core> [cfg/cfg_script.c:80]: ERROR: new_cfg_script_var(): variable already exists: my.route_addr Aug 18 12:04:56 appliance /usr/local/sbin/kamailio[2771]: ERROR: cfg_db [cfg_db.c:335]: cfg_db: Error declaring cfg int 'my.route_addr' Aug 18 12:04:56 appliance /usr/local/sbin/kamailio[2771]: ERROR: <core> [sr_module.c:860]: init_mod(): Error while initializing module cfg_db Thanks for looking. TRUONG NGOC THANH Telecommunications Engineer Tel: 0984 480 646 Y!M: ngoc217thanh

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