sr-users July 2009

sr-users@lists.kamailio.org

71 participants
105 discussions

[Kamailio-Users] Generate INVITE from Openser
by ALAEDDINE abbech 14 Aug '09

14 Aug '09

Hello all How can i generate an INVITE from OpenSER. That means that Openser make the role of UAC. Who have an idea for this pb. Thanks Regards Aladin

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[Kamailio-Users] Load Balancing Asterisk using Kamailio
by Kurt Weasel 14 Aug '09

14 Aug '09

Hi, I searched around the web to load balance asterisk servers and found Kamailio for possible solution. Let's say I have two identical asterisk servers with same dialplan and configuration and I want both servers look like they have same IP address from clients. Based on my understanding, we need 1 Kamailio as load balancer and 2 Asterisk servers as the real servers. Let's say the setup is : Kamailio load balancer 192.168.2.1 Asterisk Server #1 192.168.2.2 Asterisk Server #2 192.168.2.3 My question is, X-Lite softphone Configuration should be set to domain 192.168.2.1, right? I also want to know the step by step configuration to set kamailio as load balancer. I have not used Kamailio before. However after searching the documentation, the step (based on my understanding) is somewhat like this : 1. Install Kamailio. I will use the step by step here : http://kamailio.org/dokuwiki/doku.php/install:kamailio-1.5.x-from-svn 2. Then, using dipatcher module, I will configure it using this guide here : http://kamailio.org/dokuwiki/doku.php/asterisk:load-balancing-and-ha Then modify the dispatcher.list file to match the IP address of my asterisk servers : *1 sip:192.168.2.2:5060 1 sip:192.168.2.3:5060* Am I missing some steps? Do I also need to configure dialplan or any other file at Kamailio load balancer? Or those two steps basically done it all for simple load balancing configuration? Thanks for your responses. Best Regard, Kurt Weasel.

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[Kamailio-Users] HOWTO rewrite to: header
by alexander merkulov 12 Aug '09

12 Aug '09

here is correct reqwest and reqwest via kamailio is there any way to rewrite To: field in kamailio because now it nto work correct reqwest(asterisk) RECEIVING FROM: 69.70.173.195:5060 CANCEL sip:34249@193.110.78.12:8484 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.180:5060;branch=z9hG4bK4c97bd4a From: "Unknown" <sip:Unknown@voip1.bravotelecom.com<sip%3AUnknown(a)voip1.bravotelecom.com> >;tag=as366a0ba0 To: <sip:34249@193.110.78.12:8484> Call-ID: 322db75c71be4ba45a61c7666b93e6be(a)voip1.bravotelecom.com CSeq: 102 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 incorrect reqwest(multi-homed kamailio, asterisk originate to internal eth) RECEIVING FROM: 69.70.173.195:5060 CANCEL sip:34249@193.110.78.12:8484 SIP/2.0 Via: SIP/2.0/UDP 69.70.173.195;branch=z9hG4bK1b1b.3c42cea6.0 From: "Unknown" <sip:Unknown@69.70.173.195 <sip%3AUnknown(a)69.70.173.195> >;tag=as6075df60 Call-ID: 4d784fa972c4e9c9795032417f46cb26(a)voip1.bravotelecom.com To: <sip:34249@192.168.2.170 <sip%3A34249(a)192.168.2.170>> CSeq: 102 CANCEL Max-Forwards: 70 asterisk Content-Length: 0 -- Merkulov Alexander

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nat_uac_test mode parameter 16
by Andrew O. Zhukov 05 Aug '09

05 Aug '09

I just spend a lot of time to fix a trouble with Cisco AS5300 (AS5350) BYE message disapearing. The same not resolved case here: http://lists.iptel.org/pipermail/serusers/2005-July/021698.html The source of trouble was in the nat_uac_test parameter 16: Here is the part of cisco log: Jul 31 12:16:00.726 UKR: CCSIP-SPI-CONTROL: act_idle_connection_created: Connid(1) created to 193.28.184.13:5060, local_port 50058 --Skipped--- Jul 31 12:16:00.738 UKR: Sent: INVITE sip:3039100@193.28.184.13:5060 SIP/2.0 Via: SIP/2.0/UDP 193.28.184.21:5060 From: <sip:2308228@193.28.184.21>;tag=12E601F8-10C7 To: <sip:3039100@193.28.184.13> Via header the same in the all availible IOS-es. As the result openser considerate the Cisco is out of the nat box. The question is: Can someone show me the case when this parameter is necessary, before I'll switch it of on the production server? Thanks in advance. Andrew O. Zhukov.

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[Kamailio-Users] $hdr error if used with an avp
by catalina oancea 04 Aug '09

04 Aug '09

Hello I need to get the value of a header using a dynamic variable. I have the name of the header in an avp and I get the value using $hdr($avp(s:headername))). This worked in kamailio 1.4.0, I am sure of it because I tried it now on 1.4.0. But now in kamailio 1.5.1 I get an error when trying to retrieve the value: /usr/sbin/kamailio[32761]: ERROR:core:pv_get_spec_name: name type is PV_NAME_OTHER - cannot resolve /usr/sbin/kamailio[32761]: ERROR:pv:pv_get_hdr: invalid name Does anybody have any idea what the problem might be? Thanks

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[Serusers] IPTel web pages still relevant?
by SIP 03 Aug '09

03 Aug '09

A brief look at the IPTel pages show pretty much nothing but comment spam for pages and pages. Does anyone even bother to maintain the pages anymore? I'm looking for more relevant documentation to SER 2.0 (or even the SIP Router project). The docs in the doc subtree of the source for both of these projects is outdated and no longer valid with respect to any information stored in the database. It still references subscriber tables and the like. All 0.9.X format DB data. Is there an update somewhere? N.

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[Kamailio-Users] Kamailio OSP-module compile problem
by Martin Werne 31 Jul '09

31 Jul '09

Hi! I have problem compiling the OSP-module in Kamailio 1.5.x OS: Ubuntu Server 8.04LTS Compiling and installing OSPToolkit is OK. The problem occures when it comes to compiling the OSP-module in Kamailio: make[1]: Entering directory `/usr/src/kamailio/modules/osp' make[1]: Leaving directory `/usr/src/kamailio/modules/osp' make[1]: Entering directory `/usr/src/kamailio/modules/osp' Compiling destination.c Compiling globals.c Compiling orig_transaction.c In file included from orig_transaction.c:40: sipheader.h:48: error: expected declaration specifiers or '...' before 'OSPTCALLID' orig_transaction.c: In function 'ospLoadRoutes': orig_transaction.c:84: error: 'OSPE_DEST_PROT' undeclared (first use in this function) orig_transaction.c:84: error: (Each undeclared identifier is reported only once orig_transaction.c:84: error: for each function it appears in.) orig_transaction.c:84: error: expected ';' before 'protocol' orig_transaction.c:85: error: 'OSPE_DEST_OSP_ENABLED' undeclared (first use in this function) orig_transaction.c:85: error: expected ';' before 'enabled' orig_transaction.c:150: error: 'protocol' undeclared (first use in this function) orig_transaction.c:154: error: 'OSPE_DEST_PROT_SIP' undeclared (first use in this function) orig_transaction.c:157: error: 'OSPE_DEST_PROT_H323_LRQ' undeclared (first use in this function) orig_transaction.c:158: error: 'OSPE_DEST_PROT_H323_SETUP' undeclared (first use in this function) orig_transaction.c:159: error: 'OSPE_DEST_PROT_IAX' undeclared (first use in this function) orig_transaction.c:163: error: 'OSPE_DEST_PROT_UNDEFINED' undeclared (first use in this function) orig_transaction.c:164: error: 'OSPE_DEST_PROT_UNKNOWN' undeclared (first use in this function) orig_transaction.c:170: error: 'enabled' undeclared (first use in this function) orig_transaction.c:174: error: 'OSPE_OSP_FALSE' undeclared (first use in this function) orig_transaction.c:189: error: 'OSPC_SOURCE' undeclared (first use in this function) orig_transaction.c: In function 'ospRequestRouting': orig_transaction.c:260: error: 'OSPTCALLID' undeclared (first use in this function) orig_transaction.c:260: error: 'callids' undeclared (first use in this function) orig_transaction.c:282: error: too many arguments to function 'ospGetCallId' orig_transaction.c:325: error: 'OSPC_E164' undeclared (first use in this function) make[1]: *** [orig_transaction.o] Error 1 make[1]: Leaving directory `/usr/src/kamailio/modules/osp' make: *** [modules] Error 2 Has tried with both OSPToolkit 3.5.2 and 3.3.6. And with both Kamailio 1.5.1 and 1.5.2 Same problem. Does anyone know what is the problem? Regards, Martin

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[Kamailio-Users] About dlg_isflagset in onreply_route
by Min Wang 31 Jul '09

31 Jul '09

Hi all In kamailio 1.5.2, I used dlg_isflagset inside a ONREPLY_ROUTE. ////////////////scripts////////////////// Route[1]{ ... Dlg_manage(); Dlg_setflag("1"); ... } onreply_route[1] { #dlg_get("$ci", "$ft", "$tt"); #xlog("L_INFO", " Dialog: $dlg(h_id) . $dlg(h_entry)\n"); if (dlg_isflagset("1")) { xlog("L_INFO", " here is it \n"); } } ///////////////////////////////////////// The dlg_isflagset seems do not match anything until there is dlg_get("$ci", "$ft", "$tt"); before it. Questions are: (1) Is it the correct usage or a pontential bug. (2) How dialog is cheched/matched in the onreply_route internally? Kind Regards Min Wang

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[Kamailio-Users] Parallel forking problem
by Dubravko Caric 30 Jul '09

30 Jul '09

Hi all, could you please give me suggestions how to simultaneously send call to PSTN destination for UAC that is logged in. So, when call comes for sip:bob@example.com it must also go to its PSTN destination 1234(a)10.0.0.2 (avp in mysql) and ring both destinations simultaneously. Example below doesn't work and I keep ending up with two INVITEs to PSTN GW and proxy doesn't fork call to UAC. in this case, if I understand correctly, this is proper behavior because append_branch() has its function only if UAC isn't logged in or on 480,486... And how to handle this when user is logged in? if (!lookup("location")) { ... } if(...) { avp_pushto("$ru", "$avp(s:PSTN_fwd)"); append_branch(); } Thanks a lot /dubravko

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[Kamailio-Users] [ANN] Ruby XCAPClient 1.0 released
by Iñaki Baz Castillo 30 Jul '09

30 Jul '09

For those interested in building SIP and SIMPLE rich presence applications in Ruby language: http://xcapclient.rubyforge.org/ http://dev.sipdoc.net/projects/ruby-xcapclient/wiki/ Ruby XCAPClient -------------------------------------------------- A XCAP client library written in Ruby language. XCAP (RFC 4825) is a protocol on top of HTTP which allows a client (usually a VoIP/SIP device) to manipulate the contents of Presence Information Data Format (PIDF) based presence documents. These documents are stored in a server in XML format. Ruby xcapclient library implements the XCAP protocol in client side, allowing the applications to get, store, modify and delete XML documents in the server. == Features * Fetch, create/replace and delete a document. * Fetch, create/replace and delete a document element (XML node). * Fetch, create/replace and delete an element attribute. * Full configurable parameters allowing customized fields for each XCAP application, such auid, XML namespace, MIME-Type, scope (:user or :global) and default document name ("index" if not set). * Fetch the namespaces and prefixes of a document node as they are used in the server. * Manage of multiple documents per XCAP application. * Fetch the XCAP server auids, extensions and namespaces ("xcap-caps" application). * SSL. * Digest and Basic HTTP authentication. * Raise custom Ruby exception for each HTTP error response. -------------------------------------------------- Enjoy. -- Iñaki Baz Castillo <ibc(a)aliax.net>

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sr-users July 2009