sr-users November 2008

sr-users@lists.kamailio.org

97 participants
171 discussions

[Kamailio-Users] irc devel meeting summary
by Daniel-Constantin Mierla 11 Nov '08

11 Nov '08

Hello, sorry for delay... but better later than never, so I am trying to summarize the outcome of the IRC devel meeting from last Thursday, Nov 06, 2008. The log is available at: http://www.kamailio.org/dokuwiki/doku.php/development:irc-meeting-minutes-0… Regarding the 1.3.x series, we set the 2oth of November as the date for 1.3.4 release. There are several fixes and this branch is still used in many production environments, so it worths investing time in it. There are some updates in 1.4.x to be ported and some issues to be investigated. For a new 1.4.x release (the 1.4.3) we look for the beginning of December as the time frame for setting the date of the release. The 1.5.x rises in the first phase the question of a shorter time release, as we have lot of new features already implemented, see: http://www.kamailio.org/dokuwiki/doku.php/features:new-in-1.5.x http://www.kamailio.org/dokuwiki/doku.php/roadmap:1.5.x Unfortunately some of the new features are not completed yet (e.g., htable, carrierroute updates, PV migration) so we will stick to 6-8 months release cycle. The next major one that will use the common layer (core and tm) from sip-router could be done in a shorter time than usual. So, beginning to mid of January should be the time to freeze the code for a release date somewhere in end of January or February. An important aspect approached during the discussions was the release management and testing management. Perhaps we need to appoint a person to manage releasing process in order to have a better synchronization of the developers. As everyone agrees, testing is very important and we try to improve it. There is a testing suite included in the source tree right now, used for static testing. The idea of a dynamic testing and testing community was introduced and hopefully we can attract people and build a group of beta testers. There is a proposal of how to implement and what to take care of, probably this one has to continue on devel and users mailing lists for a proper organizational structure. GIT became as a subject of migrating from SVN, because of its features for a distributed development environment. However, sourceforge does not have such offerings, so it is not possible now anyhow. However the sip-router.org will use it. Regarding the sip-router.org project, we focused on short term goals, acknowledging that we have several modules offering same functionality but relying on different underlaying data structures, this have to co-exists for a longer time. For the first phase, core + tm the overlapping is rather small, the two projects insisting in the past to improve different aspects of them. Unfortunately I forgot to bring up some punctual topics proposed, mainly related to tm failure route calling in case of sending error and reply handling/dropping in onreply_routes. We can continue those subjects on the mailing lists. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

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[Kamailio-Users] Question regarding config file operation
by Siddhardha Garige 11 Nov '08

11 Nov '08

Hello all, I have a basic question regarding config file operation. Here is my understanding. At build time bison and flex work on config file and generates cfg.tab.c. This file is equivalent to c version of config file. Make uses cfg.tab.c along with other files and loads a series of actions into action structures. These actions will be performed on messages in a predefined order. If i make a change to config file, do I have to rebuild openser? But I do see just by stopping and starting these changes get reflected in operation. How does these changes get into action structure with out bison and flex generating cfg.tab.c file and make compiling with other files? Any help in this regard is appreciated. Thanks Sid

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[Serusers] Information request about IPv6
by Mary Allen (mrallen) 11 Nov '08

11 Nov '08

Hi, If somebody is using SERi as a Proxy with IPv6, would you please let me know the modifications/additions needed to have it working with the IPv6 and IPv4. We have three type of trunks to the Proxy (IPv4, IPv6 and Dual). IPv4 and IPv6 are configured for the machine. For the Config file do we just have to add another line for listening on one for the IPv4 and another for the IPv6? and for the routes based on the phone number I will have to entered one for an IPv6. Will that be all? Thanks, Mary

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[Kamailio-Users] OpenSER HA example with vrrp/ucarp?
by sergejf 11 Nov '08

11 Nov '08

We would like to set up two OpenSER 1.4.x servers in a failover configuration. The active (primary) server has two main roles: 1) load balancing SIP INVITEs between SBCs from various carriers using the CarrierRoute module 2) collecting CDRs from all these PSTN calls using the Acc module and storing them in a MySQL database for later billing reconciliation. Right now we're using a F5 Big-IP load balancer to handle the pool of two SIP proxies which receives the SIP INVITEs from our Media Servers and dispatches them to the SBCs. However, should the primary proxy fail, the secondary proxy would never receive (for example) the SIP BYE from an SBC. Solving that issue with the Big-IP seems complicated and a simpler vrrp/ucarp solution looks attractive. Any (hands-on) advice on how to set up vrrp/ucarp to advertise a single virtual IP address, then detect that the active SIP proxy has gone down and switch to the secondary IP, would be most welcome. We're using Centos 5.x (64 bit) on our servers. Thanks a lot. Serge -- View this message in context: http://www.nabble.com/OpenSER-HA-example-with-vrrp-ucarp--tp20443679p204436… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

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[Serusers] SIP/2.0 500 Server Internal Error using Mysql
by Nel Tulaylay 11 Nov '08

11 Nov '08

Hi Tech support, I just completed configuring my SER SIP Server, I'm experiencing an intermittent error of "SIP/2.0 500 Server Internal Error" during REGISTER and whenever im using a mysql server for authentication. Most of the time, i can register successfully to the SIP Server but there are times i encounter error "SIP/2.0 500 Server Internal Error". Do you have an idea what's causing this error. I have here a trace log of SIP Messages. I will greatly appreciate any support from you. Thanks Nel U CLIENT-IP:1384 -> SERVER-IP:5060 REGISTER sip:SERVER-IP SIP/2.0..Via: SIP/2.0/UDP 192.168.1.136:5060;br anch=z9hG4bK63235644e9b491928f5..From: 638871680007 <sip:638871680007@116.6 8.141.251>;tag=63283ff..To: 638871680007 <sip:638871680007@SERVER-IP>. .Call-ID: 321f3c-24e8a-491928f0@SERVER-IP..CSeq: 6 REGISTER..Contact: "638871680007" <sip:638871680007@192.168.1.136:5060>..Authorization: Digest username="638871680007", realm="SERVER-IP", nonce="49199956fed351b86a 5c11c83d849a62144c5dcd", uri="sip:SERVER-IP", response="6a29e5839028f9 bc690c77fda9031054", algorithm=MD5..User-Agent: VaxSIPUserAgent/3.0..Expire s: 3600..Max-Forwards: 70..Content-Length: 0.... U SERVER-IP:5060 -> CLIENT-IP:5060 SIP/2.0 100 Trying..Via: SIP/2.0/UDP 192.168.1.136:5060 ;branch=z9hG4bK63235 644e9b491928f5;received=CLIENT-IP..From: 638871680007 <sip:638871680007 @SERVER-IP>;tag=63283ff..To: 638871680007 <sip:638871680007@116.68.141 .251>..Call-ID: 321f3c-24e8a-491928f0@SERVER-IP..CSeq: 6 REGISTER..Ser ver: Sip EXpress router (0.9.6 (x86_64/linux))..Content-Length: 0..Warning: 392 SERVER-IP:5060 "Noisy feedback tells: pid=16275 req_src_ip=203.1 89.8.149 req_src_port=1384 in_uri=sip:SERVER-IP out_uri=sip:116.68.141 .251 via_cnt==1".... U SERVER-IP:5060 -> CLIENT-IP:1384 SIP/2.0 500 Server Internal Error..Via: SIP/2.0/UDP 192.168.1.136:5060 ;bran ch=z9hG4bK63235644e9b491928f5;rport=1384;received=CLIENT-IP..From: 6388 71680007 <sip:638871680007@SERVER-IP>;tag=63283ff..To: 638871680007 <s ip:638871680007@SERVER-IP>;tag=04b7610a616ee809532470b3b2dd8840.0e13.. Call-ID: 321f3c-24e8a-491928f0@SERVER-IP..CSeq: 6 REGISTER..Server: Si

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[Serusers] SIP/2.0 500 Server Internal Error using Mysql
by Nel Tulaylay 11 Nov '08

11 Nov '08

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Re: [Kamailio-Users] CDRTool problem
by Ricky Gutierrez 10 Nov '08

10 Nov '08

hello, I change the values inside the section db_class, but it follows me showing the error class DB_radius extends DB_Sql { var $Host = "localhost"; var $Database = "radius"; var $User = "radius"; var $Password = "mypassword"; var $Halt_On_Error ="no"; } class DB_radius2 extends DB_Sql { var $Host = "localhost"; var $Database = "radius"; var $User = "radius"; var $Password = "mypassword"; var $Halt_On_Error ="no"; } "ser_radius"=>array( "name" => "OpenSER", "class" => "CDRS_ser_radius", "table" => "radacct", # "table" => "radacct".date("Ym"), # "db_class" => array("DB_radius2","DB_radius"), "db_class" => "DB_radius", "rating" => "1", "normalizedField" => "Normalized", "UserQuotaClass" => "OpenSERQuota", "AccountsDBClass" => "DB_openser", "UserQuotaNotify" => "1", "purgeCDRsAfter" => 120, // how many days to keep the CDRs "SIPProxies" => array("10.0.0.1" => "node01", // friendly names in sip trace table "10.0.0.2" => "node02", ), "db_class_siponline" => "DB_openser", "mediaDispatcher" => "10.0.0.1" // where to get the active media sessions from MediaProxy 2.0 I am using the version of freeradius 1.1.3 and cdrtool 6.6.10 greetingss ________________________________ Sunday, November 9, 2008 4:45:43 PM Tiago Rés Wrote: Hello, I had the same error. The problem is in the section db_class in global.inc. The default config file has in that section an array with radius1 and radius2. You have to configure it properly. That problem didn't occur more, but now When I try to read the tables it always checks cdrtool.radacct instead of radius.radacct. I don't know why it happens but I think someone else had that problem. Hope I helped.

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[Serusers] Instant Messaging? Too Many Hops
by lin9k＠yahoo.com 10 Nov '08

10 Nov '08

Hi, How do I set up the instant messaging in my SER? When I use a SIP client to send a simple message the SER responds back "Too Many Hops". How do I make it work in my SER? Ling

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[Serusers] RTP Proxy used without NAT
by Marc Bechtel 10 Nov '08

10 Nov '08

Hi, we are testing the iptel.org service using the sip server provided by iptel (213.192.59.75). Sjphonev. 1.65 is used as client. We noticed a strange behavior concerning the RTP Proxy. All SIP messages pass the iptel sip proxy as record route is used. Also all RTP packets pass the sip/rtp proxy. The question bothering us is why the RTP packets are using a proxy? (sniffs show that all RTP packets are send to the proxy) To make our case clear I will explain some details: 1. We are definitely using public IP addresses. (no RFC 1918 addresses) 2. There is no NAT. 3. Sniffs show that in the first invite packet the SDP media attribute is set to nortpproxy:yes 4. In the Sip Header via section the "rport" attribute is used. (not sure why) As I understand the SER documentation RTP proxy should only be used, if the nat_uac_test function in the nathelper module returns true. This function is checking for RFC 1918 addresses and differences in IP and port values. Thanks a lot for the help. Marc

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[Kamailio-Users] branch route
by Alex R.S.M 10 Nov '08

10 Nov '08

Hi, I am trying to modify the header of both main and the branch routes, more specifically trying to modify FROM field on both routes to different values. The change on main route is fine but couldn't make the change on the branch side. if (is_method("INVITE")) { .... seturi("sip:1111111111@x.x.x.x:5060;transport=UDP"); // main route append_branch("sip:2222222222@y.y.y.y:5060"); // branch route if ( uri=~"sip:.+@x.x.x.x.*" ) { uac_replace_from("..."); // make changes on main route, works } if ( uri=~"sip:.+@y.y.y.y.*" ) { uac_replace_from("..."); // make changes on the branch, not working } .... When I move the logic for the branch into "branch_route" it would created two FROM fields. Is this a bug? Thanks, Alex _________________________________________________________________ Stay up to date on your PC, the Web, and your mobile phone with Windows Live http://clk.atdmt.com/MRT/go/119462413/direct/01/

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sr-users November 2008