- sr-users - kamailio.org

[Serusers] Dubt on RFC 4484 and SIP Proxy Server behavior
by Francesco la Torre 09 Sep '08

09 Sep '08

Hi all, I've a doubt regarding a statement at the end of page 5 "... Some important design decisions are associated with carrying assertions in a SIP request. If an assertion is carried by value, or uses a MIME-based content indirection system, then proxy servers will be unable to inspect the assertion themselves. If the assertion were referenced in a header, however, it might be possible for the proxy to acquire and inspect the assertion itself. There ..." Why a proxy server will be unable to inspect the assertion themself ? SER has functions by which it could be read message body, does this behavior not respect any rfc ? I'm missing something :-S Thanks in advance, Francesco la Torre

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[Kamailio-Users] [Fwd: Re: Unconditional call forward]
by Stefano Palleschi 09 Sep '08

09 Sep '08

Hi Gustavo, thanks for your reply. I've changed the *avp type* from "2" to "0" but I get always the same error: ERROR:avpops:load_avps: incomplet uri <sip:81.29.176.57> Now I'm going to reboot the server... If I'll have news I'll let you know. Regards, Stefano -------- Messaggio Originale -------- Oggetto: Re: [Kamailio-Users] Unconditional call forward Data: Wed, 3 Sep 2008 10:38:42 -0300 Da: Gustavo Mistrinelli <gmistrinelli(a)gmail.com> A: Stefano Palleschi <stefano.palleschi(a)okcom.it> CC: users(a)lists.kamailio.org Referenze: <48BE6299.5000805(a)okcom.it> Hi Stefano, Please check *avp type* that you are using, use type 0 instead of 2 See below the different types for avps: 0 - *AVP* with string name and string value 1 - *AVP* with string name and integer value 2 - *AVP* with integer name and string value 3 - *AVP* with integer name and integer value Regards, Gustavo On Wed, Sep 3, 2008 at 7:10 AM, Stefano Palleschi <stefano.palleschi(a)okcom.it <mailto:stefano.palleschi@okcom.it>> wrote: Hello all, i'm trying to do an unconditional call forward with openser. I've added the instructions below in the script: if (avp_db_load("$ru/username", "$avp(s:callfwd)")) { avp_pushto("$ru", "$avp(s:callfwd)"); xlog("$avp(s:callfwd)"); route(1); exit; } and i've created a record on the usr_preferences table of openser database: mysql> select * from usr_preferences; +------+------------+-----------------------------+-----------+-----------------------------+------+---------------------+ | uuid | username | domain | attribute | value | type | modified | +------+------------+-----------------------------+-----------+-----------------------------+------+---------------------+ | 1 | 0666620996 | 81.29.176.57 <http://81.29.176.57> | callfwd | sip:0666653218@81.29.176.57 <mailto:sip%3A0666653218@81.29.176.57> | 2 | 2008-09-03 09:51:33 | +------+------------+-----------------------------+-----------+-----------------------------+------+---------------------+ 1 row in set (0.00 sec) like showed, i would like forward the call that arrive at the sip client 0666620996 to the pstn number 0666653218. At openser restarting I get always this error: ERROR:avpops:load_avps: incomplet uri <sip:81.29.176.57 <http://81.29.176.57>> without make any call. Why in the uri there isn't the first part (username@)? Thanks in advance. Stefano Palleschi _______________________________________________ Users mailing list Users(a)lists.kamailio.org <mailto:Users@lists.kamailio.org> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users -- Gustavo Mistrinelli

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[Kamailio-Users] 302 redirect inside redirect
by Kent Pirlo 09 Sep '08

09 Sep '08

Ok, I am trying to get a redirect server working correctly, only cache is I need to hit another redirect server first, and grab info from the contact in the that 302 reply before processing and sending back a 300 to the original switch. call flow would be something like this: call comes into kamailio, 1) kamailio sends the invite to another proxy(1), 2) that proxy returns a 302,(if it times out it moves on to step 5) 3) kamailio sends an ACK back to that proxy(1), 4) kamailio looks at the contact header of the 302 reply, 5) depending on that info, I decide where to tell the originating switch where to go, 5) kamailio sends originating switch a 300 redirect with new contact header. 6) originating switch sends back ACK. I have tried doing this with a first step going to route(1) , which does a "t_on_reply" and "t_relay" to the second proxy. The onreply route, graps the contact saves it into variable and then sends to route(2), Route(2) looks at the variable, does some other lookups and sends t_reply("300","redirect"). The switch then sends the ACK, but kamailio says its not a local ACK and ignores it.. Then it keeps sending 300's back to the switch... Please help.. Thanks in Advance. kent

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[Kamailio-Users] Documentation Site Down
by Jon Farmer 09 Sep '08

09 Sep '08

Hi When I go to http://kamailio.org/docs/ i get a 404 error. Any ideas anyone? Regards Jon

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[Kamailio-Users] Forward call
by Thiago Rondon 09 Sep '08

09 Sep '08

Folks, My problem for forward call, that I need to change username/password and realm to do my forward. So, I try this: modparam("uac","from_restore_mode", "auto") modparam("uac","auth_realm_avp","$avp(s:uac_realm)") modparam("uac","auth_username_avp","$avp(s:uac_username)") modparam("uac","auth_password_avp","$avp(s:uac_password)" route { ... if(uri=~"^sip:0[0-9]{10}@") { xlog("L_INFO","uac_atual: $avp(s:uac_username) !!"); $avp(s:uac_username)="username"; $avp(s:uac_realm)="1.1.1.1"; $avp(s:uac_password)="password"; xlog("L_INFO","uac_trocado $avp(s:uac_username) !!"); route(3); exit; } .. } ... route[3] { t_on_failure("3"); # reset flag to mark no authentication yet performed resetflag(7); # forward to PSTN uac_replace_from("username","sip:username@1.1.1.1"); rewritehostport("1.1.1.1:5060"); xlog("L_INFO","rewritehost !!!!!!!!"); t_relay(); } .. So, when I do xlog, in the first entry, uac_username don't display, why ? And its strange, because in one supplier its ok, and in another its not run. But I don't undestand why uac_username is empty in the first xlog! Thanks, -Thiago Rondon

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[Kamailio-Users] populate_leg_info
by David Villasmil 08 Sep '08

08 Sep '08

Anyone lnow what this means? dialog:populate_leg_info: bad sip message or missing Contact hdr ERROR:dialog:dlg_onreq: could not add further info to the dialog openser 1.3.2 thnks

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[Kamailio-Users] New OpenXCAP release 1.0.0
by Adrian Georgescu 08 Sep '08

08 Sep '08

Hello, There is a new release of OpenXCAP server available. It contains many bug fixes and additions for important features like XML partial manipulation, xcap-caps, xcap-diff and RLS services that were not available at the initial release last year. With this features OpenXCAP fulfills now most if not all relevant requirements for an XCAP server implementation. A generic xcap client library and command line client is available in the clients directory, you can use it to manipulate documents on OpenXCAP or other XCAP server implementation. The server software is also operational at http://sip2sip.info so you can test against it by registering a SIP account. Changes in version 1.0.0 * Added RLS services (RFC 4662 and RFC 4826) * Added support for xcap-diff based on draft-ietf-simple-xcap-diff-09 * Added partial get/put/delete of elements in the XCAP documents * Added test suite for rls-services, resource-lists and partial updates * Many bug fixes from field experiences * Development status from beta to production * Switched to Python 2.5 * Improved documentation and testing suite * Changed license to GPL * Fixed MySQL operational error (2006) The software can be downloaded from: http://openxcap.org Many thanks to Denis Bilenko who did most of the works for this new release. Kind regards, Adrian Georgescu

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[Kamailio-Users] CDRTool and Prepaid
by Adrian Georgescu 08 Sep '08

08 Sep '08

Hello Thanks for the feedback from everyone. You can try cdrtool version 6.6.0, I have added parallel prepaid call support. The algorithm is described in PREPAID.txt cdrtool (6.6.0) unstable; urgency=low * Allow simultaneous calls for prepaid accounts, see the Simultaneous prepaid calls section in doc/PREPAID.txt * Added pseudo code examples for call control engine * Apply the changes from setup/mysql/alter_tables.mysql * Added interface to kill ongoing sessions in Prepaid page Testing and feedback is useful and welcome. Regards, Adrian

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[Kamailio-Users] call forward problem
by muhammad akl 06 Sep '08

06 Sep '08

hey all after making call forward feature using avpops module and its exported functions avp_db_load and avp_pushto i have a problem : assuming all calls to extension 12001 will be forwarded to extension 12008, in opensers's database i have added the following : username=12001 domain =mydomain attribute = callfwd value = sip:12008@mydomain type = 0 untill that every thing was fine and all calls comes to 12001 forward to 12008 without any problem , the problem comes from when i am using 12008 to call 12001 it calls itself ( mean 12008) so anyone can tell me what i have missed ? i have configured my openser.cfg like this in loaded modules section added loadmodules "avpops.so" > in module parameter section added : modparam("avpops", "avp_url", "mysql://openser:openserrw@localhost/openser") modparam("avpops", "avp_table", "usr_preferences") in Invite message handler added : if(avp_db_load("$ru/username","$avp(s:callfwd)")) { avp_pushto("$ru", "$avp(s:callfwd)"); route(1); exit; } any suggestions ?

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[Kamailio-Users] two proxies sharing the same database
by toly 05 Sep '08

05 Sep '08

Greetings, I have 2 proxies having the same database on backend. One is the primary, second is standby - OpenSER 1.2.1. That's how I implement DNS based failover. What I see, when all UA's registered on primary and I start standby, I's starting to cleanup location table according to it's own timer. So record is gone, when primary trying to updated the record on re-register it of course fails, as a result location table is wrong. What will be a solution? Thanks, Toly -- View this message in context: http://www.nabble.com/two-proxies-sharing-the-same-database-tp19335185p1933… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

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[Kamailio-Users] How to solve early media during serial forking
by Iñaki Baz Castillo 05 Sep '08

05 Sep '08

Hi, AFAIK there is an issue in typical SIP/PSTN enviroments in the case of failover using different gateways serially. The main issue is that SIP doesn't define how a UAC should handle the case in which it receives various "183 Session Progress" from different early-dialogs, each of them containing early-media. It occurs to me that a SIP/PSTN gateway sometimes replies 183 with early media for 1 or 2 seconds, and inmediatelly it replies a 502. Well, this 502 could be caused by other node in the PSTN network, it's not necessarialy a problem fo the gateway [1]. In this case I do serial forking forwarding the request to other gateway. But the UAC already received early media from the first gateway, and most of the SIP phones just render the first early media they receive in case of multiple 183, and drop others. So basically the problem is that the UAC user doesn't listen the ringing coming from PSTN after the request has been forwarded to the second gateway. Is there something I could do at proxy level to avoid this issue? Sincerely I don't think so, since it makes no sense to disallow the first 183 (I cannot know that the gateway will later reply a 502). Thanks. [1] Anyway, *who* is creating the early media? Why the PSTN generates it if the request doesn't arrive to the called? "imagine there is no PSTN..." -- Iñaki Baz Castillo

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[Kamailio-Users] custom routing
by toly 05 Sep '08

05 Sep '08

Greetings, Suppose I need to do custom routing for outbound calls from UA - softphone. Each user account may have the following info attached to it: 1:provider_a:2:provider_b:3:provider_c which means: forward call to provider_a, if failed forward call to provider_b if failed forward call to provider_c if failed send response back to UA I do database lookup based on from, and get above string. Suppose I wrote my own dispatcher module which initially gets set to above string: 1:provider_a:2:provider_b:3:provider_c and forwards to first provider, then on failure_route I call next_provider and so on. Will it work across multiple messages? Will it work correctly for each child - I mean joe@domain and paul@domain should be handled according their own providers? Any other suggestions? Thanks, Toly -- View this message in context: http://www.nabble.com/custom-routing-tp19322478p19322478.html Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

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[Kamailio-Users] connection to remote mysql database
by Asim Riaz 05 Sep '08

05 Sep '08

I am using Kamailio 1.2 , in t openser.cfg I defined the the url for db, which is not local ("auth_db|permissions|uri_db|usr_loc", "db_url", "mysql://openser:db_password@db_ip_address/openser") but still getting the error message below when start openser; /sbin/openser[13064]: permissions - initializing /sbin/openser[13064]: WARNING: File not found: //etc/openser/permissions.allow /sbin/openser[13064]: Default allow file (//etc/openser/permissions.allow) not found => empty rule set /sbin/openser[13064]: WARNING: File not found: //etc/openser/permissions.deny /sbin/openser[13064]: Default deny file (//etc/openser/permissions.deny) not found => empty rule set /sbin/openser[13064]: new_connection: Can't connect to local MySQL server through socket '/var/run/mysqld/mysqld.sock' (2) /sbin/openser[13064]: ERROR: group_db_bind: unable to connect to the database /sbin/openser[13064]: ERROR:group:mod_init: unable to open database connection /sbin/openser[13064]: init_mod(): Error while initializing module group /sbin/openser[13064]: INFO:mi_fifo:mi_destroy: process hasn't been created -> nothing to kill any idea ? thanks in advance Asim

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[Kamailio-Users] Openser & Centile
by BERGANZ François 05 Sep '08

05 Sep '08

Hello, I have some problem with, openser & centile! When do a call from centile to openser, the call don't do "ringing", just "trying" and after, centile say "BYE" Have you an idea? Thank you

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Re: [Kamailio-Users] Openser & Centile
by BERGANZ François 05 Sep '08

05 Sep '08

I could have a look, but Asterisk-Openseràno problems Centile-openseràdont work ! Cordialement BERGANZ François <http://www.acropolistelecom.net> http://www.acropolistelecom.net TEL/FAX : 33 (0) 1 70 72 50 15 De : Darren Sessions [mailto:dmsessions@gmail.com] Envoyé : vendredi 5 septembre 2008 15:26 À : BERGANZ François Objet : Re: [Kamailio-Users] Openser & Centile Would be helpful to see some of the SIP signaling. _____________________________ Darren Sessions dmsessions(a)gmail.com http://www.darrensessions.com _____________________________ On Sep 5, 2008, at 7:24 AM, BERGANZ François wrote: Hello, I have some problem with, openser & centile! When do a call from centile to openser, the call don't do "ringing", just "trying" and after, centile say "BYE" Have you an idea? Thank you _______________________________________________ Users mailing list Users(a)lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users

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[Kamailio-Users] CDRTool and Prepaid
by Adrian Georgescu 05 Sep '08

05 Sep '08

The problem with concurrent prepaid calls and single balance is that you have to correlate between the call control and rating angine somehow so that all calls terminate when balnce becomes zero. The problem is a bit complex: Example: Balance = 10. A call starts to destination XXX, for the sake of example max session time = 2 minutes After one minute, you start second call to destination YYY which has a different price and your balance is not anymore 10 but depends on the duration of the first call which is in progress. What is the maximum session time for it given that the first call is already in progress? What should happen with the first call? I am looking for suggestions on implementing a proper algorithm to deal with this situation in the rating engine. If you have any I would be glad to hear it. Adrian

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[Serusers] /tmp/ser_fifo disaapears after rebooting
by Joseph Myalla 05 Sep '08

05 Sep '08

I have revmoved the ser and reinstalled again, just after the installation ser works, i can see the PID, calls are made and text messages between computers, lokking in the tmp/ser_fifo is availabe, but it desappears after rebooting, what could be the reason On Wed, Sep 3, 2008 at 11:00 AM, <serusers-request(a)lists.iptel.org> wrote: > Send Serusers mailing list submissions to > serusers(a)lists.iptel.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.iptel.org/mailman/listinfo/serusers > or, via email, send a message with subject or body 'help' to > serusers-request(a)lists.iptel.org > > You can reach the person managing the list at > serusers-owner(a)lists.iptel.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Serusers digest..." > > > Today's Topics: > > 1. Error opening ser's fifo /tmp/ser_fifo (Joseph Myalla) > 2. Fwd: Error opening ser's fifo /tmp/ser_fifo sorry its ubuntu > 8.04LTs (Joseph Myalla) > 3. Re: Error opening ser's fifo /tmp/ser_fifo (Atle Samuelsen) > 4. SER 0.9.7-pre3 and timers (Steven C. Blair) > 5. how can i get the ip ? ( ?? ) > 6. how can i get the ip? ( ?? ) > 7. how can i get the IP? ( ?? ) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Tue, 2 Sep 2008 12:12:39 +0100 > From: "Joseph Myalla" <chanangae(a)gmail.com> > Subject: [Serusers] Error opening ser's fifo /tmp/ser_fifo > To: serusers(a)lists.iptel.org > Message-ID: > <531a7e5d0809020412n27b87d6g8f7346a569d98286(a)mail.gmail.com> > Content-Type: text/plain; charset=ISO-8859-1 > > I have downloades ser's debian packages and installed them in Ubuntu > Linux 6.0Ls Desktop edition > ser_0.9.7_i386.deb and the correspoding ser_mysql modules, > > trying commands serctrl ps or ul show i get an error "Error opening > ser's fifo /tmp/ser_fifo", trying to add a user using serctl add .. I > get error SER/FIFO not accessible: 2 > > the are some instructions in the mailing list saying I should add > fifo_mode=0777 in the configuration file, I have done it but still the > errors exist. > > can any one help. > > > -- > Myalla J.C > Kibaha Education Centre > P.O Box 30054 Kibaha > Pwani > Tanzania > +255787680744 > > > ------------------------------ > > Message: 2 > Date: Tue, 2 Sep 2008 12:14:24 +0100 > From: "Joseph Myalla" <chanangae(a)gmail.com> > Subject: [Serusers] Fwd: Error opening ser's fifo /tmp/ser_fifo sorry > its ubuntu 8.04LTs > To: serusers(a)lists.iptel.org > Message-ID: > <531a7e5d0809020414q25b69c15ya029023b51ce367e(a)mail.gmail.com> > Content-Type: text/plain; charset=ISO-8859-1 > > I have downloades ser's debian packages and installed them in Ubuntu > Linux 8.0Ls Desktop edition > ser_0.9.7_i386.deb and the correspoding ser_mysql modules, > > trying commands serctrl ps or ul show i get an error "Error opening > ser's fifo /tmp/ser_fifo", trying to add a user using serctl add .. I > get error SER/FIFO not accessible: 2 > > the are some instructions in the mailing list saying I should add > fifo_mode=0777 in the configuration file, I have done it but still the > errors exist. > > can any one help. > > > -- > Myalla J.C > Kibaha Education Centre > P.O Box 30054 Kibaha > Pwani > Tanzania > +255787680744 > > > > -- > Myalla J.C > Kibaha Education Centre > P.O Box 30054 Kibaha > Pwani > Tanzania > +255787680744 > > > ------------------------------ > > Message: 3 > Date: Tue, 2 Sep 2008 13:18:02 +0200 > From: Atle Samuelsen <clona(a)cyberhouse.no> > Subject: Re: [Serusers] Error opening ser's fifo /tmp/ser_fifo > To: Joseph Myalla <chanangae(a)gmail.com> > Cc: serusers(a)lists.iptel.org > Message-ID: <20080902111802.GD63335(a)mystic.cyberhouse.no> > Content-Type: text/plain; charset=us-ascii > > Hi Joseph, > > have you started ser? because, if ser is not running you will not be > able to get access to /tmp/ser_fifo. > > You can also check if it's there by doing a ls /tmp/ser_fifo :) > > > - Atle > > > * Joseph Myalla <chanangae(a)gmail.com> [080902 13:13]: >> I have downloades ser's debian packages and installed them in Ubuntu >> Linux 6.0Ls Desktop edition >> ser_0.9.7_i386.deb and the correspoding ser_mysql modules, >> >> trying commands serctrl ps or ul show i get an error "Error opening >> ser's fifo /tmp/ser_fifo", trying to add a user using serctl add .. I >> get error SER/FIFO not accessible: 2 >> >> the are some instructions in the mailing list saying I should add >> fifo_mode=0777 in the configuration file, I have done it but still the >> errors exist. >> >> can any one help. >> >> >> -- >> Myalla J.C >> Kibaha Education Centre >> P.O Box 30054 Kibaha >> Pwani >> Tanzania >> +255787680744 >> _______________________________________________ >> Serusers mailing list >> Serusers(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serusers > > > ------------------------------ > > Message: 4 > Date: Tue, 2 Sep 2008 11:14:46 -0400 > From: "Steven C. Blair" <blairs(a)isc.upenn.edu> > Subject: [Serusers] SER 0.9.7-pre3 and timers > To: "serusers(a)iptel.org" <serusers(a)iptel.org> > Message-ID: > <20C52563FC353043A5A9B80D847590E102D6824900(a)MAILBOX.exchange.upenn.edu> > > Content-Type: text/plain; charset="us-ascii" > > > I'd like to ask if anyone if running a service using SER 0.9.7-pre3 software with greater than 2000 registrations? If so are any of you experiencing issues which seem to be related to timers not firing when expected. We set the inv_timeout to a number which approximates a certain number of rings. This worked well when our subscriber base was small however as the number of subscribers, and associated calling activity, grew the timers seem to take longer to fire. We are planning on upgrading to a newer version of SER but I'd like to hear about other real world experiences. > > Thanks, Steve > > > > >

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[Kamailio-Users] Mediaproxy vs. RTPproxy.
by Alex Balashov 05 Sep '08

05 Sep '08

This must be an incredibly dumb question, but why would anyone ever use mediaproxy over rtpproxy? If one is written in Python and one in C, it seems to me that the performance and scalability makes the choice of rtpproxy a no-brainer. What benefit is there to using mediaproxy? More features and capabilities, ease of configuration perhaps? -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599

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[Kamailio-Users] FW: Call Hangup from Caller-End not working with TLS
by Ali Jawad 05 Sep '08

05 Sep '08

Hi All I am using using openser 1.3..if I make a call between two softphones on the same lan or a a pstn call to my mobile phone..and the called/receiver party does hang-up the call. It works fine in UDP mode and the call get's hang-up. However in TLS mode this does not work. Anything I might have missed here? Since both udp and tls use the same routes, and voice is fine and no one way audio ..etc. I notice this http://pastebin.com/m38c979f6 on rtp proxy. However in the logs of openser I can't see any errors. However on the wire shark I can see icmp destination unreachable...port unreachable. I would have said it is a NAT issue. However it works for simple UDP. However I did notice the following in the logs Aug 28 13:41:01 [8565] DBG:tm:set_timer: relative timeout is 4000000 Aug 28 13:41:01 [8565] DBG:tm:insert_timer_unsafe: [7]: 0xb6137794 (45900000) Aug 28 13:41:01 [8565] DBG:tm:retransmission_handler: retransmission_handler : done Aug 28 13:41:02 [8564] DBG:core:parse_msg: SIP Request: Aug 28 13:41:02 [8564] DBG:core:parse_msg: method: <BYE> Aug 28 13:41:02 [8564] DBG:core:parse_msg: uri: <sip:michofr@193.227.186.146:3218;transport=UDP;rinstance=D98C1DD404B200 8F980980E97E42F8EC;nat=yes> Aug 28 13:41:02 [8564] DBG:core:parse_msg: version: <SIP/2.0> Aug 28 13:41:02 [8564] DBG:core:parse_headers: flags=2 Aug 28 13:41:02 [8564] DBG:core:parse_via_param: found param type 232, <branch> = Shouldn't the transport=TLS ? Regards

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[Serusers] how can i get the ip?
by 邱磊 05 Sep '08

05 Sep '08

hi guys: I get a sip message from my asterisk,and I want to route the message to the ip contained in the "record-route" header.How can i get the ip and transport the message? regards 2008-09-05 邱磊

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[Kamailio-Users] Help with SerMyAdmin
by Tiago Rés 04 Sep '08

04 Sep '08

Hello all, I'm a new user in this list, but I have already seen some posts about serMyAdmin. I have openser 1.2.2 installed, running with mysql Version 14.12. I'm trying to use a web interface for openser, such as sermyadmin or openser administrator, but I'm unable to accomplish that. I have followed the instructions from sermyadmin website, but I had some errors: I couldn't add a user in mysql with that command, so I used something like: openserctl add admin openserrw admin@localhost The remaining steps went ok, but everytime I try to open http://(myserver):8080/serMyAdmin I get an error page from Tomcat. Any suggestion? Thanks in advance. Tiago

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[Kamailio-Users] Fax Detection
by Abdul Hakeem 04 Sep '08

04 Sep '08

Hello, Is there a way to identify in incoming call as a fax (perhaps by detecting CNG tone ). Regards, Abdul Hakeem

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[Kamailio-Users] ERROR:mysql:db_mysql_new_connection: driver error: Too many connections
by Samael - 04 Sep '08

04 Sep '08

Hi, We are deploying OpenSER in a large infrastructure together with Asterisk to support thousands of current transactions. We are using udp listener and mysql module in openser. We had to increase the children parameter of the openser.cfg to 200 because Openser couldn't handle the high amount of concurrent SIP REQUESTs from ASTERISK (Not only INVITES, but also a lot of SIP OPTIONs), that means that Openser couldn't handle more than 'children' REQUEST loosing the rest of the concurrent REQUEST. children=200 But then, the following error appeared in the openser: ERROR:mysql:db_mysql_new_connection: driver error: Too many connections This is because the Standard parameter of mysql "max_connections" is set to 100. In mysql CLI, we can see that every child process of openser has a mysql connection. (see in Mysql> SHOW PROCESSLIST) The recommended max_connections parameter is 500-1000 => we have a limit of 500-1000 concurrent SIP REQUEST to be processed by the openser. Why there is a process for every child? Why is neccesary a mysql connection for every child? Can the openser have a pool of mysql connection to be used by thousands of children? Regards

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[Kamailio-Users] OpenSER reply 513 when register use FQDN
by Steven Wu 04 Sep '08

04 Sep '08

Hi, I just want use FQDN in my networking environment. My SJPhone use FQDN register to OpenSER 1.2.2 but was rejected by 513 Message to big. However it can be registed use the same profile just replace FQDN to IP address. The DNS query process worked fine and got the right IP address. My environment elments looks like: OpenSER 1.2.2 at host 192.168.1.120 / ipcsip.org DNS server at host 10.57.39.53 SJPhone at host 192.168.1.105 REGISTER package looks like: 192.168.1.105 -> 192.168.1.120 REGISTER sip:ipcsip.org:5062 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.105;branch=z9hG4bKc0a801690000001f48bf8e4700006b9400000002;rport From: "unknown" <sip:1000@ipcsip.org:5062>;tag=328715c58a2 To: <sip:1000@ipcsip.org:5062> Contact: <sip:1000@192.168.1.105> Call-ID: 444CD1265B43407EBBF2B26BD4C01C6B0xc0a80169 CSeq: 1 REGISTER Max-Forwards: 70 User-Agent: SJphone/1.65.377a (SJ Labs) Content-Length: 0 192.168.1.120 -> 192.168.1.105 SIP/2.0 513 Message too big Via: SIP/2.0/UDP 192.168.1.105;branch=z9hG4bKc0a801690000001f48bf8e4700006b9400000002;rport=5060 From: "unknown" <sip:1000@ipcsip.org:5062>;tag=328715c58a2 To: <sip:1000@ipcsip.org:5062>;tag=2560d490c3265ff35995c6bbde62a7c3.b7d8 Call-ID: 444CD1265B43407EBBF2B26BD4C01C6B0xc0a80169 CSeq: 1 REGISTER Server: OpenSER (1.2.2-notls (i386/linux)) Content-Length: 0 I got same result evenif enabled dns & rev_dns in openser.cfg (dns=yes rev_dns=yes). So, What wrong with me? Coudle some give me some instructions? Best regards, Steven Wu

4 7

[Kamailio-Users] Unconditional call forward
by Stefano Palleschi 04 Sep '08

04 Sep '08

Hello all, i'm trying to do an unconditional call forward with openser. I've added the instructions below in the script: if (avp_db_load("$ru/username", "$avp(s:callfwd)")) { avp_pushto("$ru", "$avp(s:callfwd)"); xlog("$avp(s:callfwd)"); route(1); exit; } and i've created a record on the usr_preferences table of openser database: mysql> select * from usr_preferences; +------+------------+-----------------------------+-----------+-----------------------------+------+---------------------+ | uuid | username | domain | attribute | value | type | modified | +------+------------+-----------------------------+-----------+-----------------------------+------+---------------------+ | 1 | 0666620996 | 81.29.176.57 | callfwd | sip:0666653218@81.29.176.57 | 2 | 2008-09-03 09:51:33 | +------+------------+-----------------------------+-----------+-----------------------------+------+---------------------+ 1 row in set (0.00 sec) like showed, i would like forward the call that arrive at the sip client 0666620996 to the pstn number 0666653218. At openser restarting I get always this error: ERROR:avpops:load_avps: incomplet uri <sip:81.29.176.57> without make any call. Why in the uri there isn't the first part (username@)? Thanks in advance. Stefano Palleschi

2 1

[Serusers] Which kind of SIP message can I use to exchange Data out of session?
by Peter Cai 04 Sep '08

04 Sep '08

Hi All, If a SIP client has to exchange some data out of a Session with the proxy server, such as get his "buddy list", what SIP Message can I use? I didn't find any thing suit for this purpose. Should I use something beside SIP such as HTTP.

3 3

[Serusers] cannot load module acc.so
by 静于 04 Sep '08

04 Sep '08

hello, I cannot load the acc.so module.The config file is below: # ------------- version 0.8.11-0 # ------------- Initial global variables debug=3 fork=yes log_stderror=no listen=10.1.2.3 listen=127.0.0.1 # hostname matching an alias will satisfy the condition uri==myself". alias=bigu.edu alias=10.1.2.3 # dns - Uses dns to check if it is necessary to add a "received=" field # to a via. Default is no. # rev_dns - Same as dns but use reverse DNS. dns=no rev_dns=no port=5060 children=4 # check_via - Turn on or off Via host checking when forwarding replies. # Default is no. arcane. looks for discrepancy between name and # ip address when forwarding replies. check_via=yes # syn_branch - Shall the server use stateful synonym branches? It is # faster but not reboot-safe. Default is yes. syn_branch=yes # memlog - Debugging level for final memory statistics report. Default # is L_DBG -- memory statistics are dumped only if debug is set high. memlog=3 # sip_warning - Should replies include extensive warnings? By default # yes, it is good for trouble-shooting. sip_warning=yes # fifo - FIFO special file pathname fifo="/tmp/ser_fifo" # server_signature - Should locally-generated messages include server's # signature? By default yes, it is good for trouble-shooting. server_signature=yes # reply_to_via - A hint to reply modules whether they should send reply # to IP advertised in Via. Turned off by default, which means that # replies are sent to IP address from which requests came. reply_to_via=no # user | uid - uid to be used by the server. 99 = nobody. uid="nobody" # group | gid - gid to be used by the server. 99 = nobody. gid="nobody" # mhomed -- enable calculation of outbound interface; useful on # multihomed servers. mhomed=0 # ------------- external module loading loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/acc.so" loadmodule "/usr/local/lib/ser/modules/exec.so" loadmodule "/usr/local/lib/ser/modules/group.so" loadmodule "/usr/local/lib/ser/modules/print.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/uri.so" # ------------- tm parameters modparam("tm", "fr_timer", 12) modparam("tm", "fr_inv_timer", 24) # ------------- rr parameters # set ";lr" tag to ?�?;lr=true?�? modparam("rr", "enable_full_lr", 1) # ------------- accounting parameters modparam("acc", "log_missed_flag", 3) modparam("acc", "log_level", 1) modparam("acc", "log_flag", 1) # ------------- usrloc parameters # 2 enables write-back to persistent mysql storage for speed # disable=0, write-through=1 modparam("usrloc", "db_mode", 2) # minimize write back window - default is 60 seconds modparam("usrloc", "timer_interval", 10) # database location modparam("usrloc", "db_url", "sql://ser:>password>@localhost/ser") # ------------- auth parameters # database location modparam("auth_db", "db_url", "sql://ser:>password>@localhost/ser") # allows clear text passwords in the mysql database modparam("auth_db", "calculate_ha1", yes) # name of password column in mysql database modparam("auth_db", "password_column", "password") # ------------- routing logic route { # ------------- routine checks # stop forwarding at 10 hops to prevent infinite loops if (!mf_process_maxfwd_header("10")) { log(1, "LOG: Too many hops\n"); sl_send_reply("483", "Too many hops"); break; }; # prevents private ip space from being used if (search("^(Contact|m): .*(a)(192\.168\.|10\.|172\.16)")) { if (method=="REGISTER") { log(1, "LOG: Someone trying to register from private IP\n"); sl_send_reply("479", "Please don't use private IP addresses" ); break; }; }; # separate the destination r-uri from the set of proxies that must be traversed loose_route(); # if the host portion of the request uri is not local, send it directly # to route processing. if (!(uri==myself)) { route(2); break; }; # All REGISTER attempts are processed and must always be authenticated if (method=="REGISTER") { # make sure that users don't register infinite loops if (search("^(Contact|m): .*(a)(10\.1\.2\.3|(proxy\.)?bigu\.edu)")) { log(1, "LOG: alert: someone trying to set aor==contact\n"); sl_send_reply("476", "No Server Address in Contacts Allowed" ); break; }; # challenge/response if (!www_authorize("bigu.edu", "subscriber")) { www_challenge("bigu.edu", "0"); break; }; # only registered users are allowed if (!is_user("replicator") & !check_to()) { log(1, "LOG: unregistered user registration attempt\n"); sl_send_reply("403", "Only registered users are allowed"); break; }; # it is an authenticated request, update Contact database now if (!save("location")) { sl_reply_error(); }; break; }; # process traffic local to BigU and the PSTN # Find the canonical username lookup("aliases"); # check domain again, if it is not still local after the alias # table lookup, just send it on its way. We do not authenticate # traffic we forward if (!(uri=~"^sip:(.+@)?(10\.1\.2\.3|(proxy\.)?bigu\.edu)([:;\?].*)?$")) { route(5); break; }; # now check for destinations through the gateway. 911 and 9911 # are always sent to the gateway. The assumption is that other all # numeric usernames between 5 and 20 digits are really pstn numbers # and so they are routed to the gateway if ( (uri=~"^sip:911@.*") | (uri=~"^sip:9911@.*") | (uri=~"sip:[0-9]{5,20}@.*") ) { route(3); break; }; # does the user wish redirection on no availability? (i.e., is he # in the voicemail (ser->grp) group?) if (is_user_in("Request-URI", "voicemail")) { t_on_failure("4"); setflag(4); }; # handle local SIP destinations not found in usrloc db # mostly offline or non-existent users if (!lookup("location")) { route(4); break; }; # check whether some inventive user has uploaded gateway # contacts to usrloc to bypass authorization logic if (uri=~"@10\.1\.2\.5([;:].*)*" ) { log(1, "LOG: Gateway address in UsrLoc\n"); route(3); break; }; # this flag is used with the acc module to report missed calls # to syslog. setflag(3); # do it (words to live by) append_hf("P-hint: USRLOC\r\n"); if (!t_relay()) { sl_reply_error(); break; }; } /* end of initial routing logic */ # ------------- process traffic leaving BigU for Internet route[2] { # outbound requests are allowed only for registered BigU users if (!(src_ip==10.1.2.3) & !(proxy_authorize("bigu.edu", "subscriber"))) { # ACK and CANCEL have no security mechanisms so they are just # noted if (method=="ACK" | method=="BYE") { log(1, "LOG: failed outbound authentication for ACK granted\n"); } else if (method=="CANCEL") { log(1, "LOG: failed outbound authentication for CANCEL granted\n"); } else { proxy_challenge("bigu.edu", "0"); break; }; }; # to maintain credibility of our proxy, we check >From in INVITEs if (!src_ip==10.1.2.3 & method=="INVITE" & !check_from()) { log(1, "LOG: Spoofed from attempt\n"); sl_send_reply("403", "Use From=id next time"); break; }; append_hf("P-hint: OUTBOUND ON INTERNET\r\n"); if (!t_relay()) { sl_reply_error(); break; }; } # ------------- process traffic leaving Internet for PSTN route[3] { # all calls through the gateway must be record routed to assure # acl acceptance on the gateway record_route(); # send out emergency calls to pstn gateway immediately if ( (uri=~"^sip:911@.*") | (uri=~"^sip:9911@.*") ) { rewritehostport("10.1.2.5:5060"); forward(uri:host, uri:port); break; }; # five digit numeric addresses are internal freebies sent to the pbx # without authentication if (uri=~"^sip:[0-9]{5}@(10.1.2.3|(proxy\,)?\.bigu\.edu)") { rewritehostport("10.1.2.5:5060"); forward(uri:host, uri:port); break; }; # all numeric addresses beginning with 9 go to the pbx on the way # to the PSTN # first the caller needs to be authenticated if (uri=~"^sip:9[0-9]*@(10.1.2.3|(proxy\.)?bigu\.edu)") { if (!(src_ip==10.1.2.3 | method==ACK | method=="CANCEL" | method=="BYE")) { if (!proxy_authorize("bigu.edu", "subscriber")) { proxy_challenge( "bigu.edu","0"); break; } else if (method=="INVITE" & !check_from()) { log(1, "LOG: Spoofed from attempt\n"); sl_send_reply("403", "Use From=id next time"); break; }; }; if (method=="INVITE") { # if the r-uri begins 91, does the authenticated user have # permission for long distance if (uri=~"sip:91[0-9]*@.*") { if (!is_user_in("credentials", "ld")) { sl_send_reply("403", "Local calls only"); break; }; }; }; # authenticated and authorized, now accounting is set setflag(1); }; rewritehostport("10.1.2.5:5060"); append_hf("P-hint: GATEWAY\r\n"); if (!t_relay()) { sl_reply_error(); break; }; } # ------------- process calls for users offline route[4] { if (!t_newtran()) { sl_reply_error(); }; if (!t_reply("404", "Not Found")) { sl_reply_error(); }; break; } # ------------- process aliased outbound traffic # inbound requests that have been aliased to a non-BigU domain # are not authenticated by BigU route[5] { append_hf("P-hint: ALIASED-OUTBOUND\r\n"); if (!t_relay()) { sl_reply_error(); break; }; } # ------------- CC-Diversion to voicemail failure_route[4] { append_branch("sip:80000@10.1.2.5"); append_urihf("CC-Diversion: ", "\r\n"); append_hf("P-hint: OFFLINE-VOICEMAIL\r\n"); t_relay(); } Can anyone help BR Ivy ___________________________________________________________ �Ż��䣬��䣡 http://cn.mail.yahoo.com/

1 0

Re: [Kamailio-Users] svn diff
by Vadim Berezniker 04 Sep '08

04 Sep '08

Oops hit Ctrl+Enter svn diff http://openser.svn.sourceforge.net/svnroot/openser/tags/1.2.2/ http://openser.svn.sourceforge.net/svnroot/openser/tags/1.2.3/ I've been trying to run this against the public SVN but it never completes.... I hate sourceforge. From: Vadim Berezniker Sent: Wednesday, September 03, 2008 1:57 PM To: 'users(a)lists.kamailio.org' Subject: svn diff Can someone with access to the dev svn run this for me?

3 3

[Kamailio-Users] Could you please help me how to integrate RLS AS and PS AS with SERIMS?
by mahesh.peddi＠info-spectrum.com 04 Sep '08

04 Sep '08

Hi, I am working on openser 1.3.2 as RLS application server and presence application server. I want to integrate this two application servers with SERIMS. Could you please help me how to integrate and make run this. To do this I have run SERIMS components on two servers? Regards, Mahesh Peddi Infospectrum India Pvt. Ltd. Cell: +91 9765775176 IP-Phone Ext. - 764

1 0

[Kamailio-Users] SIP-T: Multipart SDPs, ISUP parsing, etc
by Kristian Kielhofner 03 Sep '08

03 Sep '08

Hello everyone, Feel free to set $OSSPROXY to whatever you like = Kamailio/OpenSIPS/OpenSER (for those of you who can't let go). $OSSPROXY is my favorite SIP toolkit. In addition to being an excellent proxy and registrar it serves as a platform to do SIP mangling and wrangling. With that in mind let me propose a crazy idea... How hard would it be to get $OSSPROXY to be able to parse what is commonly referred to as SIP-T (RFC 3372)? I understand $OSSPROXY has no business getting too involved in the SDP but I feel it makes sense in some instances (like this one). What if I want to make a routing decision based on an encapsulated ISUP parameter of some sort? Who knows what it could be!?!? Maybe there are some other uses but that makes the most sense and seems to be the most practical. It should be possible to develop an $OSSPROXY module to: - Use the SDP parser to find the ISUP SDP part: http://www.kamailio.net/dokuwiki/doku.php/development:sdp-parsing - Pass that part into a "real" ISUP parser (if necessary). I couldn't seem to find the perfect library but perhaps something like this could work: http://www.openss7.org/isup.html - Make the ISUP parameters available via AVPs, pseudo variables, etc for use in the $OSSPROXY script. I don't really have a need for this but it seems like it could be cool. Am I nuts? -- Kristian Kielhofner http://blog.krisk.org

3 3

[Serusers] Is setdsturi supported in SER
by Steven C. Blair 03 Sep '08

03 Sep '08

I am wondering if setdsturi is supported in SER 0.9.7-pre3 Thanks

1 0

[Kamailio-Users] svn diff
by Vadim Berezniker 03 Sep '08

03 Sep '08

Can someone with access to the dev svn run this for me?

1 0

[Kamailio-Users] Unconditional call forward
by Stefano Palleschi 03 Sep '08

03 Sep '08

Hello all, i'm trying to do an unconditional call forward with openser. I've added the instructions below in the script: if (avp_db_load("$ru/username", "$avp(s:callfwd)")) { avp_pushto("$ru", "$avp(s:callfwd)"); xlog("$avp(s:callfwd)"); route(1); exit; } and i've created a record on the usr_preferences table of openser database: mysql> select * from usr_preferences; +------+------------+-----------------------------+-----------+-----------------------------+------+---------------------+ | uuid | username | domain | attribute | value | type | modified | +------+------------+-----------------------------+-----------+-----------------------------+------+---------------------+ | 1 | 0666620996 | 81.29.176.57 | callfwd | sip:0666653218@81.29.176.57 | 2 | 2008-09-03 09:51:33 | +------+------------+-----------------------------+-----------+-----------------------------+------+---------------------+ 1 row in set (0.00 sec) like showed, i would like forward the call that arrive at the sip client 0666620996 to the pstn number 0666653218. At openser restarting I get always this error: ERROR:avpops:load_avps: incomplet uri <sip:81.29.176.57> without make any call. Why in the uri there isn't the first part (username@)? Thanks in advance. Stefano Palleschi

2 1

[Kamailio-Users] VQ_MALLOC memory manager
by Henning Westerholt 03 Sep '08

03 Sep '08

Hi all, does somebody uses the VQ_MALLOC memory manager? I thought about removing this one, as it don't work correctly at the moment (it lacks a realloc implementation). This would be a good cleanup in the core. If somebody still depends on this memory manager, perhaps has also some fixes created for this, please reply to this thread. If nobody speaks up i'll remove the VQ_MALLOC implementation at the end of this month. Cheers, Henning

1 0

[Kamailio-Users] NAT transversal without mediaproxy or rtpproxy
by BERGANZ François 03 Sep '08

03 Sep '08

Hello, I am looking for to do nat transversal without mediaproxy or rtpproxy! I cant find any example without mediaproxy or rtpproxy. Have you an idea? I need to register my users, and contact them after Thank you

4 6

[Kamailio-Users] t_relay to two diffrent proxy
by Asim Riaz 03 Sep '08

03 Sep '08

Hi , I am trying to relay invite from Kamailio to one of two proxies, as long as it receive the reply from one it should not forward to second but incase if the proxy1 does not reply then relay to proxy2. here is my config route(1); if((src_ip != proxy1_ip_address) || (src_ip !=proxy2_ip_address) ) { if(!t_relay_to_udp("proxy1_ip_address", "5060")) || (!t_relay_to_udp("proxy2_ip_address","5060")) { end_media_session(); sl_reply_error(); } } I am also allowing only my proxies can see the location tables; f ((src_ip== Proxy1_ip_address) || (src_ip= Proxy2_ip_address)) { if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; for some reason incoming calls from my proxies are looping means Kamailio sending back to one of my proxyies, i dont know if there is somthing missing in my configuration or somthing else is causing this problem. Regards Asim Riaz

2 2

[Kamailio-Users] Kamailio dispatcher Module redirection
by Asim Riaz 03 Sep '08

03 Sep '08

Hi All I am trying to configure Openser as loadblancer for two more Kamailio. I wan loardbalancer not stay in the middle once routed to other kamilio servers. my first question is, dispathcer list always take IPs or I can configure proxy names as well. at the moment I am checking the to_uri and then redirecting the traffic to my one of Kamilio on my network, it seems to work fine with Linksys but not other phones e.g polycom xlite or even sjphone. if(to_uri=~"sip:.+@sip.mydomain.ie <sip%3A.%2B(a)sip.mydomain.ie>") { ds_select_domain("1", "4"); sl_send_reply("300","Redirect"); exit; } any suggestion. Thanks in advance Regards Asim Riaz

4 8

[Kamailio-Users] uac_redirect question
by Zahid Mehmood 03 Sep '08

03 Sep '08

Hi, Need a little help with get_redirects(), using openser 1.2. I am using the following in failure_route to capture contacts from a "multiple choices" reply from cisco media gateway. Following code sets the ruri to the last contact, however, the $ds show more information about other contacts. Is there a way to reset this so that only ruri information is used? Cisco response and xlog information is also given below. if(!get_redirects("*:*")) { xlog("L_ERROR", "Failed to fetch contact '$ct' from 301/302 - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci \n"); acc_db_request("480", "acc"); t_reply("480", "Temporarily Unavailable"); exit; } # get last URI from destination-set and set it as R-URI xlog("L_INFO", "Redirect from UAC intercepted 1 - M= $rm RURI=$ru D=$ds B=$bR \nF=$fu T=$tu IP=$si ID=$ci\n"); avp_delete("$avp(s:tmp)/g"); $avp(s:tmp) = $ds; avp_subst("$avp(s:tmp)", "/.*(sip:.+@[^:;>]+).*$/\1/"); avp_pushto("$ru", "$avp(s:tmp)"); setflag(29); xlog("L_INFO", "Redirect from UAC intercepted 2 - M= $rm RURI=$ru D=$ds B=$bR \ntmp=$avp(s:tmp) \nF=$fu T=$tu IP=$si ID=$ci \n"); append_branch(); route(17); # process exit; } Thanks in advance for your advice. -- Zahid The response from gateway: U 2008/08/26 08:19:04.440411 10.10.0.32:5060 -> 10.10.0.98:5060 SIP/2.0 300 Multiple Choices. Via: SIP/2.0/UDP 10.10.0.98;branch=z9hG4bK901b.335cb695.0,SIP/2.0/UDP 10.10.12.140;branch=z9hG4bK612a4f20B9293665. From: "10521" <sip:10521@devproxy.myip.org>;tag=B395F9F2-4618AC3F. To: <sip:40001@devproxy.myip.org;user=phone>;tag=195535C4-11AC. Date: Tue, 26 Aug 2008 12:19:04 GMT. Call-ID: f7932e23-d89cd014-e3ea20a9(a)10.10.12.140. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 2 INVITE. Allow-Events: telephone-event. Diversion: <sip:40001@10.10.0.32>;reason=unconditional;counter=1. Contact: <sip:10512@10.10.0.98>,<sip:10512@10.10.0.32>. Content-Length: 0. xlog entries: Aug 26 08:19:04 mousse openser[15353]: Redirect from UAC intercepted 1 - M=INVITE RURI=sip:40001@10.10.0.32:5060;transport=udp D=Contact: sip:40001@10.10.0.32:5060;transport=udp, <sip:10512@10.10.0.32 >;q=0.01, <sip:10512@10.10.0.98>;q=0.01 B=<sip:10512@10.10.0.32>;q=0.01, <sip:10512@10.10.0.98>;q=0.01 F=sip:10521@devproxy.myip.org T=sip:40001@devproxy.myip.org;user=phone IP=10.10.12.140 ID=f7932e23-d89cd014-e3ea20a9(a)10.10.12.140 Aug 26 08:19:04 mousse openser[15353]: Redirect from UAC intercepted 2 - M=INVITE RURI=sip:10512@10.10.0.98 D=Contact: sip:10512@10.10.0.98, <sip:10512@10.10.0.32>;q=0.01, <sip:10512@10.10.0.98 >;q=0.01 B=<sip:10512@10.10.0.32>;q=0.01, <sip:10512@10.10.0.98>;q=0.01 tmp=sip:10512@10.10.0.98 F=sip:10521@devproxy.myip.org T=sip:40001@devproxy.myip.org;user=phone IP=10.10.12.140 ID=f7932e23-d89cd014-e3ea20a9(a)10.10.12.140 Aug 26 08:19:04 mousse openser[15353]: Redirect from UAC intercepted - M=INVITE RURI=sip:10512@10.10.0.98 D=Contact: sip:10512@10.10.0.98, <sip:10512@10.10.0.32>;q=0.01, <sip:10512@10.10.0.98 >;q=0.01, sip:10512@10.10.0.98 B=<sip:10512@10.10.0.32>;q=0.01, <sip:10512@10.10.0.98>;q=0.01, sip:10512@10.10.0.98 F=sip:10521@devproxy.myip.org T=sip:40001@devproxy.myip.org;user=phone IP=10.10.12.140 ID=f7932e23-d89cd014-e3ea20a9(a)10.10.12.140

2 1

[Kamailio-Users] t_relay
by sabino frisardi 03 Sep '08

03 Sep '08

Hi all, I'm using openser 1.4. Can anyone tell me how can i call t_relay() from another module. Can anyone help me? any suggestion? Thanks for your help. Regards -- Sabino Frisardi

3 2

[Serusers] how can i get the IP?
by 邱磊 03 Sep '08

03 Sep '08

hi guys: I get a sip message from my asterisk,and I want to route the message to the ip contained in the "record-route" header.How can i get the ip and transport the message? 2008-09-03 邱磊

1 0

[Serusers] how can i get the ip?
by 邱磊 03 Sep '08

03 Sep '08

hi guys: I get a sip message from my asterisk,and I want to route the message to the ip contained in the "record-route" header.How can i get the ip and transport the message? regard~ 2008-09-03 邱磊

1 0

[Serusers] how can i get the ip ?
by 邱磊 03 Sep '08

03 Sep '08

hi guys: I get a sip message from my asterisk,and I want to route the message to the ip contained in the "record-route" header.How can i get the ip and transport the message? regard~ 2008-09-03 邱磊

1 0

Re: [Kamailio-Users] CDRTool and Prepaid
by Matteo D'Amato 03 Sep '08

03 Sep '08

Thanks David for your answer, but can you explain how radius can do that since sending BYE messages is more at the signaling level. I dont quite understand how radius can even communicate at the SIP signaling level. From: David Villasmil Sent: Sunday, August 31, 2008 4:33 PM To: Matteo D'Amato Subject: Re: [Kamailio-Users] CDRTool and Prepaid You could use radius. On Sun, Aug 31, 2008 at 10:09 PM, Matteo D'Amato > wrote: Hello, In "/doc/PREPAID.txt" it talks about "an external call control module (not provided by CDRTool) that maintains call status and terminates calls by sending BYEs to both SIP end-points" What is this 3-party module and where can I get it? --Matteo _______________________________________________ Users mailing list Users(a)lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users

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[Serusers] SER 0.9.7-pre3 and timers
by Steven C. Blair 02 Sep '08

02 Sep '08

I'd like to ask if anyone if running a service using SER 0.9.7-pre3 software with greater than 2000 registrations? If so are any of you experiencing issues which seem to be related to timers not firing when expected. We set the inv_timeout to a number which approximates a certain number of rings. This worked well when our subscriber base was small however as the number of subscribers, and associated calling activity, grew the timers seem to take longer to fire. We are planning on upgrading to a newer version of SER but I'd like to hear about other real world experiences. Thanks, Steve

1 0

[Serusers] Error opening ser's fifo /tmp/ser_fifo
by Joseph Myalla 02 Sep '08

02 Sep '08

I have downloades ser's debian packages and installed them in Ubuntu Linux 6.0Ls Desktop edition ser_0.9.7_i386.deb and the correspoding ser_mysql modules, trying commands serctrl ps or ul show i get an error "Error opening ser's fifo /tmp/ser_fifo", trying to add a user using serctl add .. I get error SER/FIFO not accessible: 2 the are some instructions in the mailing list saying I should add fifo_mode=0777 in the configuration file, I have done it but still the errors exist. can any one help. -- Myalla J.C Kibaha Education Centre P.O Box 30054 Kibaha Pwani Tanzania +255787680744

2 1

[Serusers] Fwd: Error opening ser's fifo /tmp/ser_fifo sorry its ubuntu 8.04LTs
by Joseph Myalla 02 Sep '08

02 Sep '08

I have downloades ser's debian packages and installed them in Ubuntu Linux 8.0Ls Desktop edition ser_0.9.7_i386.deb and the correspoding ser_mysql modules, trying commands serctrl ps or ul show i get an error "Error opening ser's fifo /tmp/ser_fifo", trying to add a user using serctl add .. I get error SER/FIFO not accessible: 2 the are some instructions in the mailing list saying I should add fifo_mode=0777 in the configuration file, I have done it but still the errors exist. can any one help. -- Myalla J.C Kibaha Education Centre P.O Box 30054 Kibaha Pwani Tanzania +255787680744 -- Myalla J.C Kibaha Education Centre P.O Box 30054 Kibaha Pwani Tanzania +255787680744

1 0

[Serusers] Local number portability
by inge 02 Sep '08

02 Sep '08

Hi all, Is someone know if it's possible to use SER for prefix calls for LNP ? Then the call will be sent to AS5350 and the prefix must define the carrier for out ported number. I search information on this topic but I find nothing about SIP and LNP. Thanks for our informations. Regards

1 0

[Kamailio-Users] route number
by BERGANZ François 02 Sep '08

02 Sep '08

Hello, I need to have a route number like route(1000) But after some tests, I cant have more than route(39) How can I have more? Thank you

2 1

[Kamailio-Users] How to config OpenSER as IMS server
by Steven Wu 01 Sep '08

01 Sep '08

Hi everybody, Currently we are working on an IMS project. This project need a IMS server. Finally we choiced OpenSER as the IMS server. But have no experience how to use OpenSER as a IMS server. Could someone tell me how to config it? Best regards, Steven Wu

3 2

[Kamailio-Users] Mediaproxy 2.0 - Dispatcher vs Relay
by Nuno Marques 01 Sep '08

01 Sep '08

Hi, I'm trying to understand better how mediaproxy works. Just read the mediaproxy-ng.org page... i understood the necessity of using mediaproxy for NAT traversal, but i have one doubt... - What's the difference between the dispatcher and the relay? Can someone explain me the concept or point me to some literature so i can clear my mind :) Thanks in advance, Nuno

3 2

[Kamailio-Users] kamailio 1.5 and beyond
by Henning Westerholt 01 Sep '08

01 Sep '08

Hi all, i want to share some thoughts where i think the priorities for the next major release(s) of Kamailio should be. This is the continuation of the work i (and many others) have done for the previous releases, but i thought it makes sense to summarize it a bit on a higher (abstraction) level. 1. Refactoring and cleanup We still have quite a lot of old code in our repository, which is no longer working or obsolete. Some implementations uses way to much gotos, or macros that are several pages long. This makes important parts of the core and modules harder to understand and maintain then necessary. Especially in the area of utility functions there is still to much duplicated code. proposed actions: - identification and removal of no longer working code in the core and modules - identification and unification of redundant (duplicated) code in core and modules to one implementation - identification and removal of obselete code parts in core and modules. - refactoring of code to use less gotos and macros 2. User and developer documentation With regards to user documentation quite a lot of work has been done in the past releases. But with the developer documentation, especially in certain modules, there is still plenty of content missing. As the involvement of individuals and companies can change, documentation is necessary to ensure a continued maintenance of the code. proposed actions: - conversion of already existing documentation to doxygen format - add a short doxygen module definition to every module, explaining the purpose - add a short doxygen file definition to every file, explaining the purpose - extend doxygen docs in important areas like the core and modules like TM - autogeneration of database related module documentation from the XML scheme definitions 3. Consolidation in core and modules At the moment we've 85 modules in our repository. Some modules implement only one or two functions, which makes no real sense with regards to the maintainance overhead that comes with it. 24 of the modules are using database access and 7 implement database interfaces. There is no clear boundary in some areas of the core and modules, this hinders the further development of the server. proposed actions: - integrate several modules that share responsibilities and not implement that much functionality on their own into one - autogeneration of module interface and database infrastructure code from the XML database scheme definitions - identify obselete module functions that can be done easily with pseudo-variables now. Remove these functions. - move certain core parts that do not need to be there into modules, making the core smaller and more flexible - identify core APIs which are used from modules, evaluate eventual generalisations to make the modules less dependent from the actual core implementation 4. Quality assurance The complexity of several modules, and the special dependencies that some of them carry increase the work to test the server and reproduce bugs. Without proper test coverage maintenance work is slowed down and flexibility is lost. All "easy" bugs that get caught automatically will not reach the user, increasing the quality. proposed actions: - extend available test suite to cover more of core and modules - continue maintenance of available infrastructure for build tests (build bot, debian package autobuilder), evaluate extensions to increase coverage All this work will be done as in the past in an open and accountable way on the developer list. For bigger changes a request for comment will be posted in advance on this list. If existing functionality is deprecated, then this RFC will be also posted to the user list. If some functionality is moved or changed than this will be documented in the wiki (porting docs) as usual. I understand some changes will be probably add work on your side, in porting, configuration adjustments and the like. But i think this is necessary to ensure the further development of the Kamailio project, and to be able to deal with new challenges that will come up in the future. Any input and help with this regards to this individual tasks is of course much appreciated. In the end the success of the project depends in the future (like it did in the past) on your support. Without your involvement in contributions, discussions and bug reports i would be only to a much lesser extend involved in this project. Thank you, Henning Westerholt -- Henning Westerholt - Development Consumer Products / DSL Core 1&1 Internet AG, Ernst-Frey-Str. 9, 76135 Karlsruhe, Germany

2 2

[Kamailio-Users] About Changing 401 Message
by Ali Jawad 31 Aug '08

31 Aug '08

Hi All When I do logon with a wrong username/pwd I get the following error 401 UNAUTHORIZED. If I do want to change that error "for whatever stupid reason :-)" I know I could change that message using the header files and recompile. However I only know where to change the UNAUTHORIZED keyword, what about the 401 number...apart from that could I change that message using ss_reply ? Thanks With Regards Ali Jawad System Administrator Splendor Telecom (www.splendor.net) Beirut, Lebanon Phone: +961 1 373725 Fax: + 961 1 375554

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Re: [Kamailio-Users] CDRTool and Prepaid
by Matteo D'Amato 31 Aug '08

31 Aug '08

Hello, In "/doc/PREPAID.txt" it talks about "an external call control module (not provided by CDRTool) that maintains call status and terminates calls by sending BYEs to both SIP end-points" What is this 3-party module and where can I get it? --Matteo

1 0

Re: [Kamailio-Users] [Kamailio-Business] project foundation
by Daniel-Constantin Mierla 31 Aug '08

31 Aug '08

Hello, On 08/13/08 08:37, Max van Biezen wrote: > Hi, > > >> a quick operational expenses estimation for the first year brought up >> the figure of ~3000Euro, that includes the expenses for creating the >> foundation, two hosted servers (one main, one backup), domain names and >> trademark registrations. The costs regarding TM registration expenses >> (US&EU) are not yet very clear to me, maybe someone can shed some light >> if he/she did it recently. >> > > I can only give an indication on EU trademark. That is 850,- one time > fee. This fee is due regardless the fact some company will complain or > not. As happened to me. > > The process can be done by yourself no lawyers needed. However there are > lawyers outhere that simply complain to everything in the hope some > company which they claim to represent willl pay for it. Anyway worst > case scenario would then be to drop the registration whihc means the 850 > is gone > > Bottom line just try it yourself. It is a long process before > registration is final (more than 6 months) > thank you for the details. Cheers, Daniel > Max van Biezen > > > -- Daniel-Constantin Mierla http://www.asipto.com

1 1

[Kamailio-Users] issue when adding subscriber to mysql
by Alex R.S.M 30 Aug '08

30 Aug '08

Hi, Ijust added mysql module to my openser 1.2 and tried to test it by adding a couple of SIP users to the database. I can verified there is only one default sub in the database: mysql> select count(*) from subscriber; +----------+ | count(*) | +----------+ | 1 | +----------+ 1 row in set (0.00 sec) mysql> show tables; I can add the first subscriber: # openserctl add 100 100 100@localhost -e database engine 'MYSQL' loaded -e Control engine 'FIFO' loaded -e is_user: user counter=0 -e check_db_alias: alias counter=0 -e mysql_query -e insert -e 0 -e new user '100' added Howver, When I try to add the second subscriber I am getting the following error: # openserctl add 101 101 101@localhost -e database engine 'MYSQL' loaded -e Control engine 'FIFO' loaded -e is_user: user counter=0 -e check_db_alias: alias counter=0 ERROR 1062 (23000) at line 1: Duplicate entry '' for key 3 -e mysql_query -e insert -e 0 -e \E[37;31mERROR: introducing the new user '101' to the database failed root@bling1:/etc/openser# What might be the problem? Thanks, Alex _________________________________________________________________ See what people are saying about Windows Live. Check out featured posts. http://www.windowslive.com/connect?ocid=TXT_TAGLM_WL_connect2_082008

2 1

[Serusers] SER Version
by Arun Kumar 30 Aug '08

30 Aug '08

Hi All, I'm planning to roll out a new production SER for my company needs suggestion on SER version. At present I run 0.9.6 and having some issue like waiting transaction, extra connection to mysql, so which version should I roll out. thanks arun

4 9

[Kamailio-Users] openser children parameter
by toly 29 Aug '08

29 Aug '08

Greetings, What would be value of children parameter for few thousands user? Thanks, Toly -- View this message in context: http://www.nabble.com/openser-children-parameter-tp19192868p19192868.html Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

3 4

[Serusers] AVP and MySQL question
by Steven C. Blair 29 Aug '08

29 Aug '08

I have a SER 0.9.7-pre3 system. I use avopops to load flags from mysql to implement call andling features. I'd like to read a list of values for a single attribute from the database (and I can today) however the order in which the results are return is not consistent. I read that MySQL4 has an issue where results of a query are returned in a nondeterministic fashion. I think I can get my desired result by issuing my own query but I do not see an avpops function which allows for a user specified query. Does such a function exist? If so what is it's name? Thanks

2 1

[Kamailio-Users] Question regarding outbound configuration.
by Siddhardha Garige 29 Aug '08

29 Aug '08

Hello all, I am working on an outbound proxy configuration and here is my setup. UA ------> Outbound ------->firewall(NAT) -------->Proxy. Both UA and outbound are behind the NAT. I did some RFC study on outbound configuration for routing calls both inbound and outbound through outbound proxy. According to RFC 3327 Extension Header field for Registering Non-Adjacent contacts, PATH variable has to be introduced into REGISTRATIONS. Here are few questions and any help with answers is greatly appreciated. 1. Default openser.cfg will route request to destination if (URI != myself) and appends p-hint:outbound. What exactly is the significance of P-hint:outbound? http://tools.ietf.org/html/draft-ietf-sip-outbound-15 says ob parameter has to be added to PATH filed or Contact Field to retain path. P-hint got to do any thing regarding this? 2. I enabled path.so module. But openser does not at PATH filed to outgoing requests. Can someone direct me to a config example? Thanks in advance, Sid

1 1

[Kamailio-Users] event notifications and user aliases
by Alexander 'Leo' Bergolth 28 Aug '08

28 Aug '08

Hi! I'd like to implement P2P presence and extension monitoring for my SIP network. Our users register by using a numeric extension (e.g. sip:226050@wu-wien.ac.at), but most of the time, they are contacted using a user alias that looks like their email-address (e.g. sip:bergolth@wu-wien.ac.at). So the addresses stored in their buddy-lists won't be the "real"-SIP-URI of the users. Will there be a problem if I translate those aliases on subscription, so that a user may also subscribe for the presence of a user alias? If I rewrite SUBSCRIBE requests to use the "real" SIP-URI on subscription, the following NOTIFY messages will contain the wrong URI in the From-header and in the pidf body. But maybe this won't be a problem, since the watcher can assign it to the correct username because the NOTIFY belongs to the same dialog as the SUBSCRIBE? Will the user agents be able to correctly assign those notifications to their contacts? Any experiences with this? Cheers, --leo -- e-mail ::: Leo.Bergolth (at) wu-wien.ac.at fax ::: +43-1-31336-906050 location ::: Computer Center | Vienna University of Economics | Austria

4 4

[Serusers] Call forwarding: Diversion header
by caio 28 Aug '08

28 Aug '08

Hi, Does anyone know how to implement draft-levy diversion header on for example bilnd call forwarding? I have this scenario: xlite ------ SER-0.9.7 ------ PSTN-GW ------ PSTN Case: Call from xlite-579950 to 579917 (both registered at SER), 579917 has fwdblind enabled to pstn-501866. where SER modifies INVITE when detect that user 579917 has fwdblind feature enabled. The call must be forwarded to the pstn number. but the PSTN-GW returns to SER a "302: Moved Temporarily" with a confused diversion header.. Here a few sip messages: = SER -----> PSTN-GW U +0.000824 200.2.127.187:5060 -> 200.2.127.153:5060 INVITE sip:501866@200.2.127.153:5060 SIP/2.0. Record-Route: <sip:200.2.127.187;ftag=2d0ad32d;lr=on>. Via: SIP/2.0/UDP 200.2.127.187;branch=z9hG4bK396d.c60b3015.0. Via: SIP/2.0/UDP 192.168.132.22:35318;branch=z9hG4bK-d8754z-c16e4506aa72e334-1---d8754z-;rport=35318. Max-Forwards: 16. Contact: <sip:579950@192.168.132.22:35318>. To: "579917"<sip:579917@wsa.lab>. From: "579950"<sip:579950@wsa.lab>;tag=2d0ad32d. Call-ID: MzI2MjQ4ZTMyMGQzOWRlYzNkMWFkZTVmNGY3MzIzNDE.. CSeq: 2 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. User-Agent: X-Lite release 1100l stamp 47546. Content-Length: 385. = PSTN-GW ----> SER U +0.051198 200.2.127.153:5060 -> 200.2.127.187:5060 SIP/2.0 302 Moved Temporarily. Via: SIP/2.0/UDP 200.2.127.187;branch=z9hG4bK396d.c60b3015.0. Via: SIP/2.0/UDP 192.168.132.22:35318;branch=z9hG4bK-d8754z-c16e4506aa72e334-1---d8754z-;rport=35318. To: "579917" <sip:579917@wsa.lab>;tag=121881197764d164f2-9431-47bb-be95-8af6684ebc04. From: "579950" <sip:579950@wsa.lab>;tag=2d0ad32d. CSeq: 2 INVITE. Call-ID: MzI2MjQ4ZTMyMGQzOWRlYzNkMWFkZTVmNGY3MzIzNDE.. Contact: <sip:579917@200.2.127.187:5060>. Diversion: <sip:501866@200.2.127.153:5060>;reason=unknown. Server: Epygi Quadro SIP User Agent/v4.1.23 (SOHOGW). Content-Length: 0. What is wrong? How the flow must be to satisfies diversion of levy draft? Any comments are welcome...i'm confused if SER must modify INVITE or "To" header or if a diversion must be set (add_diversion of diversion.so module) when SER send the first invite to PSTN-GW... Regards, -- caio

1 0

[Kamailio-Users] Why dialplan and drouting modules were removed from openser-1.3.3?
by Peter Lemenkov 28 Aug '08

28 Aug '08

Hello All! Subj. -- With best regards!

3 4

[Kamailio-Users] How to test RLS and trace messages?
by mahesh.peddi＠info-spectrum.com 28 Aug '08

28 Aug '08

Hi, I am working on RLS Application server and integrated with SER IMS, i made changes in the configuration file with the help of Readme file. Its working fine. But how to confirm and test RLS is working. And how to trace the messages? Regards, Mahesh Peddi Infospectrum India Pvt. Ltd. Cell: +91 9765775176 IP-Phone Ext. - 764

1 0

[Kamailio-Users] Using pua_bla for BLF
by catalina oancea 28 Aug '08

28 Aug '08

Hello I am trying to make some phones work with BLF using Kamailio and I found an older messge ( http://lists.kamailio.net/pipermail/users/2007-October/013700.html) saying that I could do this using BLA. I have a Thomson ST2030 as the message says, but it doesn't work with presence and pua_bla. I get the error: ERROR:presence:handle_subscribe: Missing or unsupported event header field value ERROR:presence:handle_subscribe: event= dialog I undestood from that message that the event is supposed to be 'dialog', so what's the problem then? Thanks a lot Catalina

3 6

[Kamailio-Users] Fw: regarding 3rd party register request.
by srinivasa.reddy＠info-spectrum.com 28 Aug '08

28 Aug '08

Hi all, This is Srinivas, i am working on OpenIMScore and Openser( as Application Server) . from few days i stuck at third party regaester request(that is interface between SCSCF and Presence ). You said that, the registration-event package is not enabled at the IMS Core side that mean in the scscf.cfg is it? If it is could you please provide where and what i have to change and i am attaching the scscf.cfg . And regarding 2nd point Also the Openser presence server, as much I know, will be for the "presence" package. Not for the registration-event "reg" package That means openser does not support reg package or ? WITH BEST REGARDS, SRINIVAS. -------------------------------------------------------------------------------------------------------------------------------- Hi, >From your mail below, I could infer only that, the registration-event package is not enabled at he IMS Core side. Make sure, that you have done the necessary settings for that. Also the Openser presence server, as much I know, will be for the "presence" package. Not for the registration-event "reg" package. Thanks, Noble Antony T. On Wed, Aug 27, 2008 at 3:54 PM, <mahesh.peddi(a)info-spectrum.com> wrote: Hi, I have one issue regarding openser could you kindly reply if you know . I am implimented openser1.2.0 and 1.3.2 as Presence server. And i am connected it with IMS core .And for testing i was used UCTIMS client during the register with uctims client actually i need to get 4 messages. 1.Third party register request is going to presence server from scscf. 2.Notify messages coming from presence server to uctclient. 3.subscribe message must take route through the presence server. 4.publish message must take route through the presence server. But at present i am getting 3 and 4 messges corectly i am not getting 1 and 2. So please tell me what are the modifications or changes i need to do in any xml files or cfg files. Regards, Mahesh Peddi Infospectrum India Pvt. Ltd. Cell: +91 9765775176 IP-Phone Ext. - 764 Regards, Srinivasa Reddy Kadiri Software Developer Infospectrum India Pvt. Ltd. Cell: +91 9921710062 Email: srinivasa.reddy(a)info-spectrum.com IP-Phone Ext. - 759

1 0

Re: [Kamailio-Users] How to test and trace the RLS messages?
by Daniel-Constantin Mierla 28 Aug '08

28 Aug '08

Hello, please keep cc-ing to mailing lists, there are people that can answer you quicker and better. As a matter of fact, although I have studied a bit IMS docs, that happened long time ago and don't know how those messages should be routed in IMS and where the mistake can be in your routing logic. Cheers, Daniel On 08/27/08 13:24, mahesh.peddi(a)info-spectrum.com wrote: > Hi, > > I have one issue regarding openser could you kindly reply if you know . > > I am implimented openser1.2.0 and 1.3.2 as Presence server. And i am > connected it with IMS core .And for testing i was used UCTIMS client > during the register with uctims client > actually i need to get 4 messages. > 1.Third party register request is going to presence server from scscf. > 2.Notify messages coming from presence server to uctclient. > 3.subscribe message must take route through the presence server. > 4.publish message must take route through the presence server. > But at present i am getting 3 and 4 messges corectly i am not getting > 1 and 2. > So please tell me what are the modifications or changes i need to do > in any > xml files or cfg files. > > Regards, > Mahesh Peddi > Infospectrum India Pvt. Ltd. > Cell: +91 9765775176 > IP-Phone Ext. - 764 > > ----- Original Message ----- From: "Daniel-Constantin Mierla" > <miconda(a)gmail.com> > To: <mahesh.peddi(a)info-spectrum.com> > Cc: "userlist" <users(a)lists.kamailio.org> > Sent: Wednesday, August 20, 2008 10:44 PM > Subject: Re: [Kamailio-Users] How to test and trace the RLS messages? > > >> Hello, >> >> On 08/20/08 09:44, mahesh.peddi(a)info-spectrum.com wrote: >>> Hi All, >>> I am working on openser 1.3.2 version with RLS as >>> "integrated_xcap_server". Its running without erros. >>> But my query is how to test, And please send me call flow between >>> IMS components and RLS server. >> perhaps you can find some hints in IMS specifications. You need in >> first place a sip phone with rls support, afaik, there are not that >> many. >> >> Cheers, >> Daniel >> >>> Regards, >>> Mahesh Peddi >>> Infospectrum India Pvt. Ltd. >>> Cell: +91 9765775176 >>> IP-Phone Ext. - 764 >>> ------------------------------------------------------------------------ >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users(a)lists.kamailio.org >>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >>> >> >> -- >> Daniel-Constantin Mierla >> http://www.asipto.com >> >> > -- Daniel-Constantin Mierla http://www.asipto.com

1 0

[Kamailio-Users] openser v1.3.3. released
by Daniel-Constantin Mierla 27 Aug '08

27 Aug '08

Hi everybody, OPENSER v1.3.3 is out - it is a minor release of the branch 1.3, that includes the fixes since v1.3.2 committed so far- configuration file and database compatibility is preserved. OPENSER 1.3.3 is based on the latest version of branch 1.3, therefore those running 1.3.2 or earlier version in this branch are advised to upgrade. Source tarballs are available at: http://www.kamailio.org/pub/openser/1.3.3/src/ Detailed changelog: http://www.kamailio.org/pub/openser/1.3.3/ChangeLog Download via SVN: svn co https://openser.svn.sourceforge.net/svnroot/openser/branches/1.3 openser Tag for this release can be browsed at: http://openser.svn.sourceforge.net/viewvc/openser/tags/1.3.3/ Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 0

[Serusers] Radius Auth : 407 proxy auth and Get-User-Attr - crash
by Samuel Muller 27 Aug '08

27 Aug '08

Hello all, i'm using SER 2.0 with Radius accounting and authentication - based on x64 ubuntu machine. i've a problem with the Radius auth, and only for the proxy auth mechanism (method INVITE) : depending of the topology of the UAC, it crashes the SER or a there's a 407 proxy auh required. when the UAC is behind a simple NAT, i get the 407, with "preauth: credentials with realm 'sip.720.fr' not found" - i tried many configs, no way it doesn't work. but the REGISTER is ok - it's the same Radius request ? when the UAC is behind NAT but in a VPN (our customers), proxy auth is successfull even if the preauth don't find the credentials with the realm (why? it does not differ from another uac without VPN ???), but when SER requests the Radius to get the user attributes, it crashes ! in each case, there's no pbm with REGISTER requests. Radius user sample : 1010(a)sip.720.fr User-Password = "pwd" User-Name = "1010", SER-UID = "1010(a)sip.720.fr", SER-DID = "1", SER-Digest-Username = "1010", SER-Digest-Realm = "sip.720.fr", SER-Uri-User = "1010(a)sip.720.fr" Does anyone have any idea, or met already this problem ? What I have to verify ? Thanks for help, Samuel MULLER

1 0

[Kamailio-Users] multiple extensions/domains
by voipmo 27 Aug '08

27 Aug '08

I have been running asterisk servers for a few years now to host multiple organizations. I have been working on setting up OpenSER so that we can have failover and load balancing. My question is how do I keep my different organizations separate? I currently have the following registrations: 1001(a)test.example.com 1002(a)test.example.com 1001(a)another.example.com The problem is that if 1002(a)test.example.com dials 1001 it rings both domains 1001 phones. Am I going about this the right way? Is there a better method? How might I prevent calls between domains? -- View this message in context: http://www.nabble.com/multiple-extensions-domains-tp19173400p19173400.html Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

2 2

[Serusers] Help with Ser and LCR
by Jai Rangi 26 Aug '08

26 Aug '08

All, I need little help with LCR module in ser, Say these are the entries on my LCR tables, Prefix, | From_URI | Grp_id | Priority "13612499080" ;"%"; "46"; "1" "1" ;"%"; "1"; "1" "1" ;"%" "2" "2" Now any Say if someone calls 19495556666, Then ser sends the call to Server in Grp_id 1, if the grp_id=1 server does not respond, then SER sends the call to Server with grp_Id 2. Perfect, that exactly what I need. Now if some calls 13612499080", then SER will send the call to server with grp_id 46. If the call failes it should not send the call to server grp_id 1 or two. Right? In my case it is sending the call to server grp_id 1 after the call fails on server grp_id 46. How can I stop that. I want the calls for 13612499080 forwarded to only server grp_id 46 and is that fails then just terminate the call. I will appreciate if someone can provide any help or hint. Thank you, -JP

1 1

[Kamailio-Users] AVP to store user contact information
by Asim Riaz 26 Aug '08

26 Aug '08

Hi All, I am wondering if there is a way to store the user contact info into avp, relay it to existing Regitrar and store the original contact info into location through a function(which i dont know) once i received successful reply from my sip registrar. Thanks in advance Regards Asim Riaz

5 17

[Kamailio-Users] FW: | SMS FALAS | <--- did anyone else on this list get this message?
by Vadim Berezniker 26 Aug '08

26 Aug '08

Sorry this is offtopic but.. Did anyone else get this email? I'm just wondering if they scraped emails from this mailing list. -----Original Message----- From: KLIENT I NDERUAR [mailto:info@hi5.com] Sent: Tuesday, August 26, 2008 11:38 AM To: Vadim Berezniker Subject: | SMS FALAS | Klient i/e Nderuar Nese Deshironi te Ndegjoni Muzika ma te rejat ather vizitone Kosova-SMS dhe do te bindeni me muziken qe ofrojm ne. Gjithashtu per Vizitoret e Kosova-SMS ka Startu me ofert te RE. SMS FALAS MUNDSUAR NGA Kosova-SMS 60 SMS NE DIT NGA NR I YT KLIKO PER ME SHUM. --> http://www.kosova-sms.tk

1 0

[Serusers] Notice call ending
by inge 26 Aug '08

26 Aug '08

Hi all, I look for a workarround to notice the caller that the call will be hang up, by generating a beep. Is someone can help me because I don't find anything about this subject ? Thanks. Regards, Adrien.

1 0

[Kamailio-Users] Having Problem routing an ACK
by Stagg Shelton 26 Aug '08

26 Aug '08

Hello, I am having an issue routing an ACK back from a new upstream carrier. The basic situation is my openser 1.3.2 receives the INVITE. I do an alias_db_lookup, set some accounting flags, and send the call to t_relay The call makes it to the PBX, as expected, and the PBX sends a 100 trying, 180 ringing. The 180 ringing is sent to the carrier. When the incoming call is answered the PBX sends a 200OK to openser, and that 200OK is sent to the carrier. When the carrier sends the ACK back it makes it to OpenSER, but is not sent to the PBX. The PBX will send several more 200 OK attempts, but each ACK from the carrier stops at openser. From what I can tell from the wireshark packet captures, the Request- Line header in the ACK from the carrier is sip:TELNUMBER@MY_OPENSER_IPADDR I have compared these packets to similar traces from other carriers and the Request-Line header from the other carriers is sip:TELNUMBER@MY_PBX_IPADDR This new carrier is telling me that they are not doing anything wrong with the ACK. RFC3261 seems specific about the construction of the ACK, but at the same time i am getting lost going to all of the other sections trying to identify details. Basically I just need to get the ACK to the PBX. What other information can I provide with this problem. Thanks Stagg Shelton

4 20

[Serusers] ser 0.10 doucmentation
by Steven C. Blair 26 Aug '08

26 Aug '08

I was wondering if anyone has documentation on the new syntax used in SER 0.10.x. I see the following in the ser-advanced.cfg example file but I do not see these parameters documented anywhere. Thanks,Steve if ($t.fr_inv_timer) { if ($t.fr_timer) { t_set_fr("$t.fr_inv_timer", "$t.fr_timer"); } else { t_set_fr("$t.fr_inv_timer"); } } Senior Network Engineer, Information Systems and Computing Networking and Telecommunications , Suite 221A /6228 University of Pennsylvania Voice:215-573-8396 FAX:215-898-9348

6 32

[Kamailio-Users] CDRTool not applying rates
by Brian Del Shasta 25 Aug '08

25 Aug '08

I believe my rating tables are configured correctly now thanks to David Villasmil who sent me examples from his working setup. However, I am still unable to see any rates. The rating engine seems to be started and I have no problem identifying and matching calls based on specific criteria. But I still see no rates. I'm wondering if anyone knows the best place to troubleshoot the rating engine would be or if anyone has any other ideas? Thank you in advance, and thanks again David for your help!

3 7

Re: [Kamailio-Users] presence fail
by Ricky Gutierrez 25 Aug '08

25 Aug '08

in this guide, he doesn't have the module presence_xml, but he has the parameter modparam (" presence ", " force_active ", 1) http://www.kamailio.net/dokuwiki/doku.php/presence:configuration-file it is strange, I believe to be seen in the wiki a same example for openser 1.3.2 greetingss rickygm

2 1

Re: [Serusers] call forward no answer (solved)
by caio 25 Aug '08

25 Aug '08

append_branch() is missed at ser examples... On Fri, Aug 15, 2008 at 4:22 PM, caio <elcaio(a)gmail.com> wrote: > Hi all, > I can't make call forwarding on no answer works on my ser-0.9.7 installation. > Here you can see a snippet of log, where the call falls on > failure_route and then to route_1 where t_relay is called. > > Log at: http://rafb.net/p/xwTmn195.html > > The call is from sip:579951 to sip:579915(who has enabled the feature > fwdnoanswer to sip:579916. > But I do not know why the INVITE after the append_branch() is still to > R-URI: sip:579915@.... > > After the relay then the call terminates with: > > Aug 15 16:13:25 [/usr/sbin/ser] SIP Reply (status):_ > Aug 15 16:13:25 [/usr/sbin/ser] version: <SIP/2.0>_ > Aug 15 16:13:25 [/usr/sbin/ser] status: <487>_ > Aug 15 16:13:25 [/usr/sbin/ser] reason: <Request Terminated>_ > > I follow the example of ser guide.., but do not work.. > I would appreciate any comment.. > > Regards, > > -- > caiogf > -- caio

1 0

[Serusers] ser + media server
by caio 25 Aug '08

25 Aug '08

Hi all, What do you recommend for media server application to integrate wih ser-0.9.x? SEM or Asterisk? Specifically i'm talking about running services such as IVR, conference and voicemail.. Thanks your suggestions.. -- caio

1 0

[Kamailio-Users] crashing problem
by Jonathan K. Creasy 25 Aug '08

25 Aug '08

We're using Carrier route with about 53,000 routes stored in a mysql databse. We're getting this crash in production that we never had in the lab. The bt is below. Does anyone have any suggestions? nline static void free_to_params(struct to_body* tb) { struct to_param *tp=tb->param_lst; struct to_param *foo; while (tp){ foo = tp->next; <-- **** Line 75 ***** pkg_free(tp); tp=foo; } } -Jonathan #0 free_to (tb=0x817cf10) at parser/parse_to.c:75 #1 0x080daf70 in clean_hdr_field (hf=0x8a8ee18) at parser/hf.c:182 #2 0x080db05b in free_hdr_field_lst (hf=0x8a905b8) at parser/hf.c:209 #3 0x080db12d in free_sip_msg (msg=0x8193030) at parser/msg_parser.c:660 #4 0x08094695 in receive_msg ( buf=0x8168220 "SIP/2.0 183 Session Progress\r\nf: \"16512552900\"<sip:16512552900@206.80.65.90>;tag=as781b438d\r\nt: <sip:+16513401989@lefty.netlogic.net>;tag=c7ad5f5020080822192510\r\ni: 396fb801653abd6f605a12153e0e3e72@20"..., len=1271, rcv_info=0xbf82f844) at receive.c:206 #5 0x080d3e0d in udp_rcv_loop () at udp_server.c:438 #6 0x0806c8f5 in main (argc=3, argv=0xbf82fa34) at main.c:834 (gdb) backtrace full #0 free_to (tb=0x817cf10) at parser/parse_to.c:75 No locals. #1 0x080daf70 in clean_hdr_field (hf=0x8a8ee18) at parser/hf.c:182 __FUNCTION__ = "clean_hdr_field" #2 0x080db05b in free_hdr_field_lst (hf=0x8a905b8) at parser/hf.c:209 No locals. #3 0x080db12d in free_sip_msg (msg=0x8193030) at parser/msg_parser.c:660 No locals. #4 0x08094695 in receive_msg ( buf=0x8168220 "SIP/2.0 183 Session Progress\r\nf: \"16512552900\"<sip:16512552900@206.80.65.90>;tag=as781b438d\r\nt: <sip:+16513401989@lefty.netlogic.net>;tag=c7ad5f5020080822192510\r\ni: 396fb801653abd6f605a12153e0e3e72@20"..., len=1271, rcv_info=0xbf82f844) at receive.c:206 msg = (struct sip_msg *) 0x8193030 __FUNCTION__ = "receive_msg" #5 0x080d3e0d in udp_rcv_loop () at udp_server.c:438 len = 1271 buf = "SIP/2.0 183 Session Progress\r\nf: \"16512552900\"<sip:16512552900@206.80.65.90>;tag=as781b438d\r\nt: <sip:+16513401989@lefty.netlogic.net>;tag=c7ad5f5020080822192510\r\ni: 396fb801653abd6f605a12153e0e3e72@20"... tmp = <value optimized out> from = (union sockaddr_union *) 0x8a91318 fromlen = 16 ri = {src_ip = {af = 2, len = 4, u = {addrl = {2850040017, 135426432, 3213031528, 3085483479}, addr32 = {2850040017, 135426432, 3213031528, 3085483479}, addr16 = {10449, 43488, 29056, 2066, 63592, 49026, 48599, 47080}, addr = "Ñ(à©\200q\022\bhø\202¿×½è·"}}, dst_ip = {af = 2, len = 4, u = {addrl = {509300942, 0, 0, 0}, addr32 = {509300942, 0, 0, 0}, addr16 = {20686, 7771, 0, 0, 0, 0, 0, 0}, addr = "ÎP[\036", '\0' <repeats 11 times>}}, src_port = 5060, dst_port = 5060, proto = 1, proto_reserved1 = 0, proto_reserved2 = 0, src_su = {s = {sa_family = 2, sa_data = "\023ÄÑ(à©\000\000\000\000\000\000\000"}, sin = {sin_family = 2, sin_port = 50195, sin_addr = {s_addr = 2850040017}, sin_zero = "\000\000\000\000\000\000\000"}, sin6 = {sin6_family = 2, sin6_port = 50195, sin6_flowinfo = 2850040017, sin6_addr = {in6_u = {u6_addr8 = '\0' <repeats 15 times>, u6_addr16 = {0, 0, 0, 0, 0, 0, 0, 0}, u6_addr32 = {0, 0, 0, 0}}}, sin6_scope_id = 0}}, bind_address = 0x818dd90} __FUNCTION__ = "udp_rcv_loop" #6 0x0806c8f5 in main (argc=3, argv=0xbf82fa34) at main.c:834 cfg_log_stderr = 0 cfg_stream = (FILE *) 0xa179008 c = <value optimized out> r = <value optimized out> tmp = 0x8148b9c "ÿÿÿÿ" tmp_len = <value optimized out> port = <value optimized out> proto = <value optimized out> ret = <value optimized out> seed = 639236262 rfd = 4 __FUNCTION__ = "main"

4 9

Re: [Kamailio-Users] Google reference to openser users and devel Archives
by H Quintana 25 Aug '08

25 Aug '08

Hi, My temporary solution was to edit the etc/hosts file in my windows xp machine. The path is: C:\WINDOWS\system32\drivers\etc\hosts And I added the lines: 193.22.119.66 www.openser.org 193.22.119.66 openser.org to point to the Kamailio servers. Hope this helps, Humberto > Subject: Re: [Kamailio-Users] Google reference to openser > users and > devel Archives > To: users(a)lists.kamailio.org > Message-ID: > <200808251326.44077.henning.westerholt(a)1und1.de> > Content-Type: text/plain; charset="utf-8" > > On Friday 15 August 2008, Rafael Risco Gonzales-Vigil > wrote: > > This is for openser/kamailio and opensips list > administrators: > > google still has references to > http://www.openser.org/pipermail/users/ and > > http://www.openser.org/pipermail/devel/ for openser > related searchings and > > those urls does not contains old files anymore so its > dificult search > > information for learning and troubleshooting about > openser/kamailio or > > opensips... I hope someone can solve this issue for > the benefit of all > > current and future community users. > > Hi Rafael, > > we don't have access to the openser.org domain, so > there is nothing what we > can do here, sorry. All access of web and mail services to > openser.org will > go to opensips infrastructure. > > If you encounter such old links with openser.org, you can > simply > change "openser.org" to "kamailio.org", > then you'll be able to access the > content, apart from using the google archives, as Alex > pointed out. > > Cheers, > > Henning >

2 1

[Kamailio-Users] failure route configuration
by Jonathan K. Creasy 25 Aug '08

25 Aug '08

I have this block in my configuration: <excerpt from failure_route[1]> if(!cr_rewrite_uri("1", "call_id")){ t_reply("403", "$fU@$fd Not allowed"); } else { # In case of failure, re-route the request prefix("+"); t_on_failure("2"); if (is_method("INVITE")) t_on_reply("1"); # Relay the request to the gateway t_relay(); } The output in my log file: Aug 18 18:41:29 lab02b /usr/local/sbin/openser[5622]: INFO: carrierroute:carrier_rewrite_msg: uri 8909 was rewritten to sip: 8909@gateway_one Aug 18 18:41:29 lab02b /usr/local/sbin/openser[5626]: INFO: Failure Route Started Aug 18 18:41:29 lab02b /usr/local/sbin/openser[5626]: INFO: carrierroute:carrier_rewrite_msg: uri 8909 was rewritten to sip: 8909@gateway_two Aug 18 18:41:29 lab02b /usr/local/sbin/openser[5626]: ERROR: tm:t_forward_nonack: no branch for forwarding Aug 18 18:41:29 lab02b /usr/local/sbin/openser[5626]: ERROR: tm:w_t_relay: t_forward_nonack failed What am I missing that I need to do prior to calling t_relay()? -Jonathan

6 11

[Kamailio-Users] Google reference to openser users and devel Archives
by Rafael Risco Gonzales-Vigil 25 Aug '08

25 Aug '08

This is for openser/kamailio and opensips list administrators: google still has references to http://www.openser.org/pipermail/users/ and http://www.openser.org/pipermail/devel/ for openser related searchings and those urls does not contains old files anymore so its dificult search information for learning and troubleshooting about openser/kamailio or opensips... I hope someone can solve this issue for the benefit of all current and future community users. rafael Lima - Peru

3 2

[Kamailio-Users] need help on starting opensips v1.4.1
by Do Nguyen Ha 25 Aug '08

25 Aug '08

hi opensips debian:/etc/opensips# /etc/init.d/opensips start Starting opensips: opensipsListening on udp: 127.0.0.1 [127.0.0.1]:5060 udp: 192.168.1.254 [192.168.1.254]:5060 tcp: 127.0.0.1 [127.0.0.1]:5060 tcp: 192.168.1.254 [192.168.1.254]:5060 Aliases: tcp: debian:5060 tcp: localhost:5060 udp: debian:5060 udp: localhost:5060 . debian:/etc/opensips# /etc/init.d/opensips status Status of opensips: opensips is not running. could anyone tell me the tips for this Thanks Ha`

2 1

[Kamailio-Users] Why does Kamailio reply 408 when 'fr_inv_timer' expires?
by Iñaki Baz Castillo 25 Aug '08

25 Aug '08

Hi, when 'fr_inv_timer' expires Kamailio CANCELs the pending branches and reply "408 Request Timeout" to the UAC. Well, I wonder if 408 is appropiate. I think "480 User Unavailable" or "487 Cancelled" would be better: I received those replies in other maillist: "408 indicates transaction timeout. You have sent request to next hop and it has not responded back. If the gateway has not responded at all (Including 100 trying), 408 is appropriate. 480 is more of application usage meaning that the user is not available and would be appropriate if the gateway has at least send one provisional response i.e. 100, 180, 183..." "408 is used consistently to mean IP-level timeout in sending a request from one element to another. It is also defined to be returned by a UA when the time specified in "Expires" has been reached. (In theory, two different error codes should be defined for these two situations, but that has not been done.) However, in a ring-no-answer situation, it is more common for an upstream proxy to send a CANCEL to the UA/gateway, in which case the UA/gateway provides a 487 response to the INVITE." This last point suggest that Kamailio could do the following: - If the proxy has received provisional responses from callee's then not send a 408 upstream since when the proxy sends CANCEL to UAS's it will receive some 487 and would return a 487 upstream. - If the proxy hasn't received provisional responses then this is a clear "408 Request timeout". So a possibility is that Kamailio just reply a 408 in real timeouts and reply 487 when 'fr_inv_timer' expires (or just reply nothing since UAS's will generate the 487). Opinions? -- Iñaki Baz Castillo <ibc(a)aliax.net>

3 2

[Kamailio-Users] Where have the OLD Module Docs gone?
by Douglas Garstang 25 Aug '08

25 Aug '08

The module documentation for versions 1.2, 1.3 etc used to be available on the web site. Where has it gone? When you clicked on the documentation link, you used to be able to choose the version of OpenSER(Kam...OpenSIPS whatever) that you wanted. Now, your taken immediately to what appears to be the documentation for the LATEST version. Doug.

7 7

[Kamailio-Users] presence fail
by Ricky Gutierrez 25 Aug '08

25 Aug '08

Hi, lists I have some problems with the simple presence in kamailio 1.4, I add to my kamailio.cfg the following line and when restarting the it shows me errors in this it lines Aug 24 19:31:41 [21946] ERROR:core:set_mod_param_regex: parameter <force_active> not found in module <presence> Aug 24 19:31:41 [21946] CRITICAL:core:yyerror: parse error in config file, line 50, column 19-20: Can't set module parameter Aug 24 19:31:41 [21946] ERROR:core:main: bad config file (1 errors) Aug 24 19:31:41 [21946] NOTICE:presence:destroy: destroy module ... I remove of the Makefile the modul presence kamailio.cfg loadmodule "presence.so" modparam("presence|auth_db|uri_db|usrloc", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("presence", "force_active", 1) modparam("presence", "max_expires", 3600) modparam("presence", "server_address", "sip:192.168.10.1:5060") where the error can be? regards rickygm

2 1

[Kamailio-Users] preparing openser 1.3.3
by Daniel-Constantin Mierla 24 Aug '08

24 Aug '08

Hello, I propose to build a new minor release for openser 1.3 branch on the 27th of August (Wednesday). There are already a lot of fixes since 1.3.2. Probably couple of backports from 1.4 will get in until Wednesday, the rest, if it is the case, shall get in 1.3.4, probably in about another month, as we are going to maintain 2 stable branches. Perhaps 1.4.1 will be out by same time or so. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 0

[Kamailio-Users] FreeRadius patch for OpenSer1.2.2
by Nuno Marques 24 Aug '08

24 Aug '08

Hi, I'm following the examples on the book "Building Telephony Systems with OpenSer". I came to Accounting and Billing chapter using FreeRadius. Flavio (the author) says we need to apply a patch to the freeradius to "fix a problem with accounting type 15". My question is - is it still necessary to apply that patch or the new version of freeradius allready covers it? I'm having big problems applying that patch because of the key signature problems. Appreciate any help in this subject. Thanks in advance, Nuno

5 5

[Kamailio-Users] AUTO: Miodrag Radulovic is out of the office (returning 07-09-2008)
by Miodrag Radulovic 23 Aug '08

23 Aug '08

I am out of the office until 07-09-2008. Privremeno nedostupan - Out-of-Office Message. Poruka koju ste poslali je prispela i bice sacuvana, ali napominjem da sam na odmoru van zemlje do 07.09.2008. i moj pristup elektronskoj posti bice veoma ogranicen. Ukoliko se radi o potrebi za hitnom intervencijom, dostupan sam preko mobilnog telefona 065/3016576. The message that you have sent will be saved in my inbox, but please note that due to summer holliday I will be out of the country until 07.09.2008. In that period my access to my email will be very limited. In case of urgency, I am available on my mobile phone: +381 65 3016576. Miodrag Radulovic General Manager COMUTEL Note: This is an automated response to your message "Users Digest, Vol 39, Issue 83" sent on 23.8.2008 0:49:18. This is the only notification you will receive while this person is away.

1 0

Re: [Serusers] SER - Waiting Transaction
by Arun Kumar 22 Aug '08

22 Aug '08

Hi, thanks for your reply. I'll ignore this but I see so many open connection to MySQL from SER and they don't get close automatically also resulting in denial of other service using my MySQL. Thanks arun On Mon, Aug 18, 2008 at 10:12 PM, Jiri Kuthan <jiri(a)iptel.org> wrote: > ignore it > > Arun Kumar wrote: > >> Hi, >> >> >> I'm running SER 0.96 in my production for a long but there is strange >> problem , when I do serctl moni and looks for transaction it shows waiting >> transaction in () almost 400+ can some one tell me why there are so many >> transaction in waiting while I see processing only 10-20 transaction. >> >> Thanks >> arun >> >> >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> Serusers mailing list >> Serusers(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serusers >> >

4 5

[Kamailio-Users] Problem to run MediaProxy 2.0.3
by Leonardo Santana 22 Aug '08

22 Aug '08

Hi all, I´m using mediaproxy 2.0.3 on OpenSuSE 11.0. When I run the "media-dispatcher" I see this message bellow: PS.: The OpenSuSE 11.0 doesn´t have the module "libgnutls-extra.so.13" only the libgnutls-extra.so.26". ######## Aug 21 17:18:32 nana media-dispatcher[5502]: [-] Log opened. Aug 21 17:18:32 nana media-dispatcher[5502]: [-] Starting MediaProxy Dispatcher 2.0.3 Aug 21 17:18:32 nana media-dispatcher[5502]: [-] Traceback (most recent call last): Aug 21 17:18:32 nana media-dispatcher[5502]: [-] File "/usr/local/mediaproxy-2.0.3/media-dispatcher", line 46, in Aug 21 17:18:32 nana media-dispatcher[5502]: [-] from mediaproxy.dispatcher import Dispatcher Aug 21 17:18:32 nana media-dispatcher[5502]: [-] File "/usr/local/mediaproxy-2.0.3/mediaproxy/dispatcher.py", line 31, in Aug 21 17:18:32 nana media-dispatcher[5502]: [-] from mediaproxy.tls import X509Credentials, X509NameValidator Aug 21 17:18:32 nana media-dispatcher[5502]: [-] File "/usr/local/mediaproxy-2.0.3/mediaproxy/tls.py", line 12, in Aug 21 17:18:32 nana media-dispatcher[5502]: [-] from gnutls import crypto Aug 21 17:18:32 nana media-dispatcher[5502]: [-] File "/usr/lib/python2.5/site-packages/gnutls/crypto.py", line 11, in Aug 21 17:18:32 nana media-dispatcher[5502]: [-] from gnutls.validators import method_args, none, ignore, list_of, one_of Aug 21 17:18:32 nana media-dispatcher[5502]: [-] File "/usr/lib/python2.5/site-packages/gnutls/validators.py", line 9, in Aug 21 17:18:32 nana media-dispatcher[5502]: [-] from gnutls.constants import * Aug 21 17:18:32 nana media-dispatcher[5502]: [-] File "/usr/lib/python2.5/site-packages/gnutls/constants.py", line 38, in Aug 21 17:18:32 nana media-dispatcher[5502]: [-] from gnutls.library import constants Aug 21 17:18:32 nana media-dispatcher[5502]: [-] File "/usr/lib/python2.5/site-packages/gnutls/library/__init__.py", line 7, in Aug 21 17:18:32 nana media-dispatcher[5502]: [-] from gnutls.library import errors Aug 21 17:18:32 nana media-dispatcher[5502]: [-] File "/usr/lib/python2.5/site-packages/gnutls/library/errors.py", line 14, in Aug 21 17:18:32 nana media-dispatcher[5502]: [-] from gnutls.library.functions import gnutls_strerror, gnutls_alert_get Aug 21 17:18:32 nana media-dispatcher[5502]: [-] File "/usr/lib/python2.5/site-packages/gnutls/library/functions.py", line 6, in Aug 21 17:18:32 nana media-dispatcher[5502]: [-] _libraries['libgnutls-extra.so.13'] = CDLL('libgnutls-extra.so.13') Aug 21 17:18:32 nana media-dispatcher[5502]: [-] File "/usr/lib/python2.5/ctypes/__init__.py", line 348, in __init__ Aug 21 17:18:32 nana media-dispatcher[5502]: [-] self._handle = _dlopen(self._name, mode) Aug 21 17:18:32 nana media-dispatcher[5502]: [-] OSError: libgnutls-extra.so.13: cannot open shared object file: No such file or directory ##### Thanks a lot!!!

2 1

[Kamailio-Users] RFC: branches/location records control
by Daniel-Constantin Mierla 22 Aug '08

22 Aug '08

Hello, one of the most hidden parts of openser to the config is the branch processing. Behind a branch is following structure: struct branch { char uri[MAX_URI_SIZE]; unsigned int len; /* Real destination of the request */ char dst_uri[MAX_URI_SIZE]; unsigned int dst_uri_len; /* Path vector of the request */ char path[MAX_PATH_SIZE]; unsigned int path_len; int q; /* Preference of the contact among contact within the array */ struct socket_info* force_send_socket; unsigned int flags; }; I am thinking in offering a fine access to this structure in configuration file, so the admin can control much closer the branch processing. By using while and and index, checks of the attributes in branches could be done. By doing it in main route, one can reduce the overload of stateful processing, by dropping a branch before going to transaction, for example. I am looking at something similar for user location records. Such a record has many attributes, and from config is no way to access them without doing a lookup and changing r-uri/branches. There could be many benefits, for example: - being able to detect if a user calls from a registered phone or not - detecting how many contacts a user has at a moment - ability to check and, eventually, drop processing before doing changes to branches and so on... The access will be provided via pseudo-variables, perhaps the location records will be fetched in variables upon calling a function that gets an AoR as parameter. I have seen on mailing lists discussions about needs that will be solved by this new feature. Any comments? Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

4 7

[Kamailio-Users] kamailio-1.4.0 and SerMyAdmin
by luzango mfupe 22 Aug '08

22 Aug '08

Hi, Im bit cnfsd, does SerMyAdmin support Kamailio-1.4.0?. I did my configuration using SerMyAdmin 0.7-Beta.war and then 0.7-beta.war files, but everytime i pointed my browser to the IP address of my installation my Tomcat6 server return the message that SerMyAdmin does not exist!!! Can someone pls'e clarify this?? .. wbr, Lu. -- Luzango Mfupe TUUNE MOBILE Tel:0128440528/0123825710 Tshwane-RSA "...Ships are safe in harbor, but they were never meant to stay there......."

2 4

[Kamailio-Users] openser 1.2.1 errors in the log
by toly 22 Aug '08

22 Aug '08

Greetings, Proxy compiled with TLS support, tls is not turned on Routing script inclulded below. Underlying database is Sybase. The following is happening: There are 25 users having as a contact each other. Softphone is Bria. Without any calls, just subscribe/notify proxy is running out of the socket in one hour. The users are or outside coming via corp firewall, or if inside, the route is setup the way the going via corp firewall, so they are natted. Proxy is in DMS on the same network as firewall. There are no problems with nat traversal. Daniel, could you please take look at the script? Thanks a lot, Toly. this is from log: <receive.c: 206> receive_msg: cleaning up 08/15 20:12:58 29156 debug <ip_addr.c: 109> tcpconn_new: new tcp connection to: xxx.xxx.xxx.xx 08/15 20:12:58 29156 debug <tcp_main.c: 402> tcpconn_new: on port 58544, type 2 08/15 20:12:58 29156 debug <tcp_main.c: 497> tcpconn_add: hashes: 96, 45 08/15 20:12:58 29156 debug <tcp_main.c: 993> handle_new_connect: new connection: 0xb6485590 98 flags: 0002 08/15 20:12:58 29156 debug <tcp_main.c: 935> WARNING: send2child: no free tcp receiver, connection passed to the least busy one (1) this is routing script: log_facility=LOG_LOCAL2 debug=7 fork=yes log_stderror=no listen=PROXY_ENV # This is where OpenSER installed port=5070 children=20 tcp_connect_timeout=2 tcp_max_connections=8192 reply_to_via = no sip_warning = yes check_via = no dns=no rev_dns=no disable_dns_blacklist=true disable_tls = 1 listen = tls:PROXY_ENV:5061 tls_verify_server = 1 tls_verify_client = 0 tls_require_client_certificate = 0 tls_method = SSLv23 tls_certificate = "/usr/local/etc/openser/tls/user/user-cert.pem" tls_private_key = "/usr/local/etc/openser/tls/user/user-privkey.pem" tls_ca_list = "/usr/local/etc/openser/tls/user/user-calist.pem" ###########################################################33 ###########################################################33 ###########################################################33 mpath = "/usr/local/lib/openser/modules" loadmodule "uac.so" loadmodule "psybase.so" loadmodule "pike.so" modparam("pike", "remove_latency", 120) modparam("pike", "reqs_density_per_unit", 10000) modparam("pike", "sampling_time_unit", 60) loadmodule "mi_fifo.so" modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") modparam("mi_fifo", "fifo_mode", 0666 ) modparam("mi_fifo", "reply_dir", "/tmp/" ) modparam("mi_fifo", "reply_indent", "\t" ) loadmodule "xlog.so" modparam("xlog", "buf_size", 4096) modparam("xlog", "force_color", 0) loadmodule "avpops.so" modparam("avpops", "db_url", "psybase://DB_USR:DB_PWD@DB_SRV/DB_DB") modparam("avpops", "avp_table", "usr_preferences") modparam("avpops", "use_domain", 1) loadmodule "sl.so" modparam("sl", "enable_stats", 1) loadmodule "tm.so" modparam("tm", "fr_timer", 30) modparam("tm", "fr_inv_timer", 120) modparam("tm", "wt_timer", 5) modparam("tm", "delete_timer", 2) modparam("tm", "noisy_ctimer", 0) modparam("tm", "ruri_matching", 1) modparam("tm", "via1_matching", 1) modparam("tm", "unix_tx_timeout", 2) modparam("tm", "restart_fr_on_each_reply", 1) modparam("tm", "pass_provisional_replies", 0) loadmodule "maxfwd.so" modparam("maxfwd", "max_limit", 256) loadmodule "uri.so" loadmodule "uri_db.so" modparam("uri_db", "db_url", "psybase://DB_USR:DB_PWD@DB_SRV/DB_DB") loadmodule "alias_db.so" modparam("alias_db", "db_url", "psybase://DB_USR:DB_PWD@DB_SRV/DB_DB") modparam("alias_db", "user_column", "username") modparam("alias_db", "domain_column", "domain") modparam("alias_db", "alias_user_column", "alias_username") modparam("alias_db", "alias_domain_column", "alias_domain") modparam("alias_db", "use_domain", 0) loadmodule "dispatcher.so" modparam("dispatcher", "list_file", "/usr/local/etc/openser/dispatcher.list") #modparam("dispatcher", "dst_avp", "$avp(i:271)") #modparam("dispatcher", "grp_avp", "$avp(i:272)") #modparam("dispatcher", "cnt_avp", "$avp(i:273)") modparam("dispatcher", "flags", 2 ) # failover mode set flag=2 loadmodule "usrloc.so" modparam("usrloc", "timer_interval", 60) modparam("usrloc", "db_url", "psybase://DB_USR:DB_PWD@DB_SRV/DB_DB") modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", 1) modparam("usrloc", "nat_bflag", 6) loadmodule "textops.so" loadmodule "acc.so" # set the reporting log level modparam("acc", "log_level", 1) # number of flag, which will be used for accounting; if a message is # labeled with this flag, its completion status will be reported modparam("acc", "log_flag", 1 ) #modparam("acc", "log_flag", 0 ) modparam("acc", "db_flag", 1 ) modparam("acc", "log_missed_flag", 1 ) modparam("acc", "db_missed_flag", 1 ) modparam("acc", "db_table_acc", "acc") modparam("acc", "db_url", "psybase://DB_USR:DB_PWD@DB_SRV/DB_DB") modparam("acc", "log_extra", "src_ip=$si;src_user=$fU;src_domain=$fd;dst_user=$rU;dst_domain=$rd") modparam("acc", "db_extra", "src_ip=$si;src_user=$fU;src_domain=$fd;dst_user=$rU;dst_domain=$rd") modparam("acc", "report_cancels", 1) loadmodule "auth.so" modparam("auth", "nonce_expire", 300) modparam("auth", "rpid_suffix", ";party=calling;id-type=subscriber;screen=yes") modparam("auth", "rpid_avp", "$avp(s:rpid)") loadmodule "auth_db.so" modparam("auth_db", "db_url", "psybase://DB_USR:DB_PWD@DB_SRV/DB_DB") modparam("auth_db", "user_column", "username") modparam("auth_db", "domain_column", "domain") modparam("auth_db", "password_column", "password") modparam("auth_db", "password_column_2", "ha1b") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "use_domain", 0) modparam("auth_db", "load_credentials", "rpid") loadmodule "domain.so" modparam("domain", "db_url", "psybase://DB_USR:DB_PWD@DB_SRV/DB_DB") modparam("domain", "db_mode", 1) modparam("domain", "domain_table", "domain") modparam("domain", "domain_col", "domain") loadmodule "mediaproxy.so" modparam("mediaproxy","natping_interval", 10) modparam("mediaproxy","mediaproxy_socket", "/var/run/proxydispatcher.sock") modparam("mediaproxy","sip_asymmetrics","/usr/local/etc/openser/sip-asymmetric-clients") modparam("mediaproxy","rtp_asymmetrics","/usr/local/etc/openser/rtp-asymmetric-clients") loadmodule "nathelper.so" modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "rtpproxy_disable_tout", 60) modparam("nathelper", "rtpproxy_tout", 1) modparam("nathelper", "rtpproxy_retr", 5) modparam("nathelper", "received_avp", "$avp(i:801)") modparam("nathelper", "sipping_from", "sip:pinger@parusivoip.net") modparam("nathelper", "sipping_method", "OPTIONS") modparam("nathelper", "sipping_bflag", 7) loadmodule "registrar.so" modparam("registrar", "received_avp", "$avp(i:801)") modparam("registrar", "default_expires", 60) modparam("registrar", "min_expires", 50) modparam("registrar", "max_expires", 0) modparam("registrar", "default_q", 0) modparam("registrar", "append_branches", 1) modparam("registrar", "case_sensitive", 0) modparam("registrar", "received_param", "received") modparam("registrar", "max_contacts", 0) modparam("registrar", "retry_after", 0) modparam("registrar", "method_filtering", 0) modparam("registrar", "path_mode", 2) modparam("registrar", "path_use_received", 0) loadmodule "rr.so" modparam("rr", "enable_full_lr", 1) modparam("rr", "append_fromtag", 1) modparam("rr", "enable_double_rr", 1) modparam("rr", "add_username", 0) loadmodule "permissions.so" modparam("permissions", "db_url", "psybase://DB_USR:DB_PWD@DB_SRV/DB_DB") modparam("permissions", "db_mode", 1) ######################################################################## ######################################################################## ######################################################################## route { if ( is_method("REGISTER") ) { if ( !pike_check_req() ) { xlog("L_NOTICE", "TRACKING [0] ALARM - TOO MANY HITS ON REGISTER on IP=$si !!"); exit; } }; if ( is_method("INVITE") ) { if ( !pike_check_req() ) { xlog("L_NOTICE", "TRACKING [0] ALARM - TOO MANY HITS ON INVITE on IP=$si !!"); exit; } }; # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); exit; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); exit; }; xlog("L_NOTICE", "TRACKING [0] NEW REQUEST - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$si"); force_rport(); # ----------------------------------------------------------------- # Record Route Section and Acc section # ----------------------------------------------------------------- if (method=="INVITE" && nat_uac_test("19")) { record_route_preset("PROXY_ENV:5070;nat=yes"); # This is where OpenSER installed xlog("L_NOTICE", "TRACKING [0] RECORD ROUTE PRESET- M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$si"); } else if (method!="REGISTER") { xlog("L_NOTICE", "TRACKING [0] RECORD ROUTE - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$si"); record_route(); }; append_hf("P-hint: rr-enforced\r\n"); if ( search("User-Agent: Parus.*") && is_method("BYE|REFER")) { # check the transport of callee $avp(i:44) = $(ru{uri.user}); if ( $avp(i:44) != $tU ) { $avp(i:45) = $avp(i:44); } else { $avp(i:45) = $tU; } avp_db_query("exec oserIsTcpTransport @to_user='$avp(i:45)', @td='$td'", "$avp(i:11)"); xlog("L_NOTICE", "TRACKING [0] BYE FROM ASTERISK - M=$rm RURI=$ru F=$fu T=$tu TU=$tU IP=$si ID=$si AVP11=$avp(i:11) AVP44=$avp(i:44) TD=$td"); if ( $avp(i:11) == "tcp" ) { xlog("L_NOTICE", "TRACKING [0] BYE FROM ASTERISK TRANSPORT IS TCP - M=$rm RURI=$ru F=$fu T=$tu TU=$tU IP=$si ID=$si"); add_uri_param("transport=tcp"); } else if ( $avp(i:11) == "tls" ) { xlog("L_NOTICE", "TRACKING [0] BYE FROM ASTERISK TRANSPORT IS TLS - M=$rm RURI=$ru F=$fu T=$tu TU=$tU IP=$si ID=$si"); add_uri_param("transport=tls"); } else if ( $avp(i:11) == "udp" ) { xlog("L_NOTICE", "TRACKING [0] BYE FROM ASTERISK TRANSPORT IS UDP - M=$rm RURI=$ru F=$fu T=$tu TU=$tU IP=$si ID=$si"); add_uri_param("transport=udp"); } else { xlog("L_NOTICE", "TRACKING [0] BYE FROM ASTERISK TRANSPORT IS NOT TCP - M=$rm RURI=$ru F=$fu T=$tu TU=$tU IP=$si ID=$si"); } } # ----------------------------------------------------------------- # ACC Section # ----------------------------------------------------------------- if ( is_method("CANCEL|BYE|INVITE|MESSAGE") ) { setflag(1); }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- if ( is_method("BYE|CANCEL") ) { xlog("L_NOTICE", "TRACKING [0] CALL TEAR DOWN: END_MEDIA_SESSION - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$si"); end_media_session(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { xlog("L_NOTICE", "TRACKING LOOSE_ROUTE - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$si"); if (!has_totag()) { xlog("L_NOTICE", "TRACKING LOOSE_ROUTE Initial loose-routing rejected - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$si"); sl_send_reply("403", "Initial Loose-Routing Rejected"); exit; } if ( is_method("INVITE") ) { if ( !allow_trusted() ) { if (!proxy_authorize("", "subscriber")) { proxy_challenge("", "0"); xlog("L_NOTICE", "TRACKING LOOSE_ROUTE proxy authorization failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); exit; } if (!check_from()) { sl_send_reply("403", "Spoofed from-URI Detected"); xlog("L_NOTICE", "TRACKING LOOSE_ROUTE spoofed from-URI etectet - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); exit; } consume_credentials(); xlog("L_NOTICE", "TRACKING LOOSE_ROUTE proxy authorized - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$si"); }; }; if ( (nat_uac_test("19") || search("^Route:.*;nat=yes")) && is_method("INVITE|REFER|ACK") ) { setbflag(6); # NONONO fix_nated_contact() xlog("L_NOTICE", "TRACKING LOOSE_ROUTE using media proxy - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$si"); use_media_proxy(); }; route(1); exit; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if ( !is_uri_host_local() ) { if ( is_from_local() || allow_trusted() ) { xlog("L_NOTICE", "TRACKING[0] IS_FROM_LOCAL OR TRUSTED - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$si"); } else { xlog("L_NOTICE", "TRACKING[0] NOT IS_FROM_LOCAL OR NOT TRUSTED - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$si"); sl_send_reply("403", "Forbidden"); exit; } }; if ( is_method("CANCEL|BYE") ) { route(4); exit; } else if ( is_method("ACK") ) { route(1); exit; } else if ( is_method("INVITE") ) { route(3); exit; } else if ( is_method("REGISTER") ) { route(2); exit; } else if ( is_method("OPTIONS") ) { route(11); exit; } else if ( is_method("MESSAGE") ) { xlog("L_NOTICE", "TRACKING [14] ROUTE14 - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$si"); route(14); exit; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); exit; }; xlog("L_NOTICE", "TRACKING[0] calling default route(1) M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); route(1); } # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- route[1] { xlog("L_NOTICE", "TRACKING [1] - M=$rm RURI=$ru D-URI=$du F=$fu T=$tu IP=$si ID=$si"); if(isbflagset(6)) { xlog("L_NOTICE", "TRACKING [1] USING NATed reply - M=$rm RURI=$ru D-URI=$du F=$fu T=$tu IP=$si ID=$si"); t_on_reply("1"); } else { xlog("L_NOTICE", "TRACKING [1] USING Standard reply - M=$rm RURI=$ru D-URI=$du F=$fu T=$tu IP=$si ID=$si"); t_on_reply("2"); } if (!t_relay()) { xlog("L_NOTICE", "TRACKING [1] t_relay() Failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; exit; } # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- route[2] { sl_send_reply("100", "Trying"); if (!www_authorize("","subscriber")) { www_challenge("","0"); exit; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); exit; }; consume_credentials(); if (!search("^Contact:\ +\*") && nat_uac_test("19")) { fix_nated_register(); setbflag(6); setbflag(7); xlog("L_NOTICE", "TRACKING [2] Fixed nated register - M=$rm RURI=$ru D-URI=$du F=$fu T=$tu IP=$si ID=$si"); } if (!save("location")) { xlog("L_NOTICE", "TRACKING [2] Saving contact failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); sl_reply_error(); }; xlog("L_NOTICE", "TRACKING [2] Registration successful - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); } # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- route[3] { xlog("L_NOTICE", "TRACKING [3] - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); if ( !allow_trusted() ) { if (!proxy_authorize("", "subscriber")) { proxy_challenge("", "0"); xlog("L_NOTICE", "TRACKING [3] proxy authorization failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); exit; } if (!check_from()) { sl_send_reply("403", "Spoofed from-URI Detected"); xlog("L_NOTICE", "TRACKING [3] spoofed from-URI etectet - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); exit; } consume_credentials(); }; if (client_nat_test("3")) { fix_nated_contact(); setbflag(6); xlog("L_NOTICE", "TRACKING [3] FIX_NATED_CONTACT - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); }; if ( alias_db_lookup("dbaliases") ) { xlog("L_NOTICE", "TRACKING [3] callee was aliased - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); } if ( !is_domain_local("$rd") ) { xlog("L_NOTICE", "TRACKING [3] call for foreign domain M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); route(5); route(1); exit; } if ( does_uri_exist() ) { xlog("L_NOTICE", "TRACKING [3] callee was local - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); if (lookup("location")) { xlog("L_NOTICE", "TRACKING [3] LOOKUP LOCATION SUCC - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); append_hf("P-hint: usrloc applied\r\n"); remove_hf("Alert-Info"); if ( proto==tls ) { append_hf("Alert-Info: 1\r\n"); } route(5); t_on_branch("1"); route(1); exit; } }; xlog("L_NOTICE", "TRACKING [3] SOFTPHONE LOOKUP LOCATION NOT SUCC M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); # # call pstn handler # if ( route(7) ) { xlog("L_NOTICE", "TRACKING [3] call to PSTN handler successful M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); exit; # will never hit it } else { xlog("L_NOTICE", "TRACKING [3] call to PSTN handler failed M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); exit; # exit for now } # # never hit this for now, possibly in the future if # forward to different domain or so # xlog("L_NOTICE", "TRACKING[3] calling default route(1) M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); route(1); exit; } # ----------------------------------------------------------------- # PSTN call section # ----------------------------------------------------------------- route[7] { xlog("L_NOTICE", "TRACKING [37] PSTN SECTION M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); if (uri =~ "^sip:011.*@.*") { xlog("L_NOTICE", "TRACKING [37] PSTN SECTION ATTEMPT TO MAKE INTERNATIONAL CALL M=$rm RURI=$ru F=$fu T=$tu TU=$tu"); sl_send_reply("403", "Forbidden - no international calls"); return (-1); }; if ($(tU{s.len}) > 11 ) { strip(1); xlog("L_NOTICE", "TRACKING [37] PSTN SECTION MORE THEN 11 DIGITS M=$rm RURI=$ru F=$fu T=$tu TU=$tu"); sl_send_reply("403", "Forbidden - no international calls"); return (-1); } if ($(tU{s.len}) == 11 ) { strip(1); xlog("L_NOTICE", "TRACKING [37] PSTN SECTION STRIP ONE DIGIT M=$rm RURI=$ru F=$fu T=$tu TU=$tu"); } if ( ($(rU{s.len}) != 10) && ($(rU{s.len}) != 2) && ($(rU{s.len}) != 3) && ($(rU{s.len}) != 4) && ($(rU{s.len}) != 1) ) { xlog("L_NOTICE", "TRACKING [37] S.LEN != 10 or 1 or 2 or 3 or 4 M=$rm RURI=$ru F=$fu T=$tu rU=$rU IP=$si ID=$ci"); sl_send_reply("403", "Forbidden - invalid number"); xlog("L_NOTICE", "TRACKING [7] LEAVING PSTN SECTION M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); return (-1); }; if (uri =~ "^sip:[2-9]{1}[0-9]{2}[2-9]{1}[0-9]{2}[0-9]{4}@.*") { xlog("L_NOTICE", "TRACKING [37] PSTN OUTBOUND $fU -> $ruri "); avp_db_query("exec oserGetCallIdForBvox @fU='$fU', @from_domain='$fd'", "$avp(i:12)"); uac_replace_from("$avp(i:12)"); xlog("L_NOTICE", "TRACKING [37] REPLACED FROM PSTN avp = $avp(i:12[1]) $fU -> $ruri "); route(5); # select next gateway simulates NEXTONE # 1 - DEV, 2 - PROD asterisk, 3 - john's nextone, 4 - prod nextone, 5 - qa ds_select_domain("GWGROUPID", "4"); t_on_failure("2"); xlog("L_NOTICE", "TRACKING [37] DISPATCHER FORWARD PSTN $fu -> $ruri "); route(1); return (1); # will never hit this } else { xlog("L_NOTICE", "TRACKING [37] Attempt to dial invalid number $fu -> $tu"); sl_send_reply("403", "Forbidden - invalid number"); xlog("L_NOTICE", "TRACKING [37] LEAVING PSTN SECTION M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); return (-1); }; } # ----------------------------------------------------------------- # DEV / TEST section # ----------------------------------------------------------------- route[6] { xlog("L_NOTICE", "TRACKING [6] DEV / TEST SECTION M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); } # ----------------------------------------------------------------- # CANCEL or BYE handler # ----------------------------------------------------------------- route[4] { xlog("L_NOTICE", "TRACKING [4] - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); if (client_nat_test("3")) { fix_nated_contact(); xlog("L_NOTICE", "TRACKING [4] FIX_NATED_CONTACT - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); }; route(1); } # ----------------------------------------------------------------- # NAT traversal section # ----------------------------------------------------------------- route[5] { xlog("L_NOTICE", "TRACKING [5] NAT TRAVERSAL SECTION - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); if (isbflagset(6)) { if (!isflagset(8)) { setflag(8); xlog("L_NOTICE", "TRACKING [5] BFLAG 6 IS SET, USING MEDIA PROXY - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); avp_printf("$avp(i:450)", "$dd"); if (avp_check("$avp(i:450)", "eq/$src_ip/g")) { xlog("L_NOTICE", "TRACKING [5] LOOKUP SUCC, FLAGS 6 AND 7 BEHIND SAME NAT $fu -> $tu $dd $src_ip"); } use_media_proxy(); }; }; } # ------------------------------------------------------------------------ # OPTIONS Message Handling # ------------------------------------------------------------------------ route[11] { xlog("L_NOTICE", "TRACKING [11] OPTIONS - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); if(nat_uac_test("19")) { fix_nated_contact(); setbflag(6); } xlog("L_NOTICE", "TRACKING [11] - M=$rm RURI=$ru D-URI=$du F=$fu T=$tu IP=$si ID=$si"); sl_send_reply("200", "Got it"); exit; } # ----------------------------------------------------------------- # NAT reply route # ----------------------------------------------------------------- onreply_route[1] { xlog("L_NOTICE", "TRACKING ONREPLY_ROUTE[1] NAT REPLY M=$rm S=$rs D=$rr F=$fu T=$tu IP=$si ID=$ci"); if (nat_uac_test("1")) { xlog("L_NOTICE", "TRACKING ONREPLY_ROUTE[1] CLIENT_NAT_TEST FIXED NATED CONTACT M=$rm S=$rs D=$rr F=$fu T=$tu IP=$si ID=$ci"); fix_nated_contact(); }; # we are checking here for a progressing return... ie a 180 Ringing or # 183 session progress -- if this occurs we don't care from here on # about failures as a gateway is handling the call... if( status =~ "18[0-9]" ) { xlog( "L_NOTICE", "TRACKING ONREPLY_ROUTE[1] [$Tf] ORR: $ci -- SIP-$rs Reset t_on_failure()\n"); t_on_failure("0"); } else { xlog( "L_NOTICE", "TRACKING ONREPLY_ROUTE[1] [$Tf] ORR: $ci -- $rs $rr\n" ); } if ( isbflagset(6) && (status=~"(180)|(183)|2[0-9][0-9]")) { #if (!search("^Content-Length:\ +0")) { if ( $cl > 0 ) { xlog("L_NOTICE", "TRACKING ONREPLY_ROUTE[1] USE MEDIA PROXY M=$rm S=$rs D=$rr F=$fu T=$tu IP=$si ID=$ci"); use_media_proxy(); }; }; exit; } # ----------------------------------------------------------------- # standard reply route # ----------------------------------------------------------------- onreply_route[2] { xlog("L_NOTICE", "TRACKING ONREPLY_ROUTE[2] STANDARD REPLY- M=$rm S=$rs D=$rr F=$fu T=$tu IP=$si ID=$ci\n"); # we are checking here for a progressing return... ie a 180 Ringing or # 183 session progress -- if this occurs we don't care from here on # about failures as a gateway is handling the call... if( status =~ "18[0-9]" ) { xlog( "L_NOTICE", "TRACKING ONREPLY_ROUTE[2] [$Tf] ORR: $ci -- SIP-$rs Reset t_on_failure()\n"); t_on_failure("0"); } else { xlog( "L_NOTICE", "TRACKING ONREPLY_ROUTE[2] [$Tf] ORR: $ci -- $rs $rr\n" ); } exit; } # ----------------------------------------------------------------- # FAILURE route # ----------------------------------------------------------------- failure_route[1] { xlog("L_NOTICE", "TRACKING FAILURE_ROUTE [1] - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci"); if(t_check_status("408|500|503")) { if ( t_check_status("408") ) { if ( ds_next_domain() ) { xlog("L_NOTICE", "TRACKING FAILURE_ROUTE [1] TRYING NEXT GW - M=$rm S=$rs D=$rr F=$fu T=$tu IP=$si ID=$ci\n"); t_on_failure("1"); route(1); return; } } else { xlog("L_NOTICE", "TRACKING FAILURE_ROUTE [1] NOT 408 STATUS - M=$rm S=$rs D=$rr F=$fu T=$tu IP=$si ID=$ci\n"); t_reply("503", "Service not available"); return; } } } # ----------------------------------------------------------------- # FAILURE route # ----------------------------------------------------------------- failure_route[2] { # If fr_timer expires t_check_status("408") is true, although $rs is <null> if( t_check_status("408") ){ xlog( "L_NOTICE", "TRACKING FAILURE_ROUTE [2] [$Tf] FR: $ci -- TIMEOUT for Gateway $rd\n" ); } else { xlog( "L_NOTICE", "TRACKING FAILURE_ROUTE [2] [$Tf] FR: $ci -- $rs reason $rr\n" ); } # basically what we do is # 1. not worry about reasonable failures (ie busy, wrong number etc...) # 2. go to next destination gateway # 3. if no destination gateway then 503 (service unavailable) # 403 -- typically ISDN network says 'not a valid number' etc.. if( t_check_status("403") ){ xlog("L_NOTICE", "TRACKING FAILURE_ROUTE [2] [$Tf] FR: $ci -- SIP-$rs Forbidden -> ISDN Cause Code 1\n" ); end_media_session(); return; } # 408 -- timeout -- typically the end party has not answered # Since we cancel t_on_failure() on a provisional response we should not be # getting a 408 timeout from a gateway at this stage.. it will just "fall through" #if( t_check_status("408") ){ # xlog("L_NOTICE", "TRACKING FAILURE_ROUTE [2] [$Tf] FR: $ci -- SIP-$rs Timeout\n" ); # return; #} # 486 -- User Busy if( t_check_status("486") ){ xlog("L_NOTICE", "TRACKING FAILURE_ROUTE [2] [$Tf] FR: $ci -- SIP-$rs Destination Busy\n" ); end_media_session(); return; } # 487 -- Request Cancelled (usually in response to a CANCEL transaction) if( t_check_status("487") ){ xlog("L_NOTICE", "TRACKING FAILURE_ROUTE [2] [$Tf] FR: $ci -- SIP-$rs Request Cancelled\n" ); end_media_session(); return; } # ok... so at this stage we try the next gateway (unless we don't have one) # if no next gateway we bail... if( ds_next_domain() ){ #t_on_reply("1"); t_on_failure("2"); xlog( "L_NOTICE", "TRACKING FAILURE_ROUTE [2] [$Tf] FR: $ci Next gateway $fU -> $tU via $rd\n" ); route(1); return; } else { xlog( "L_NOTICE", "TRACKING FAILURE_ROUTE [2] [$Tf] FR: $ci No more gateways -> 503.\n" ); end_media_session(); t_reply("503", "Service unavailable -- no more gateways" ); return; } } # ----------------------------------------------------------------- # BRANCH route # ----------------------------------------------------------------- branch_route[1] { if (isbflagset(6)) { xlog("L_NOTICE", "TRACKING BRANCH_ROUTE[1] Branch marked as NATed- M=$rm branch=$T_branch_idx, branch flags=$bf F=$fu T=$tu IP=$si ID=$ci"); } else { xlog("L_NOTICE", "TRACKING BRANCH_ROUTE[1] Branch marked as not NATed- M=$rm branch=$T_branch_idx, branch flags=$bf F=$fu T=$tu IP=$si ID=$ci"); } } # ------------------------------------------------------------------------ # MESSAGE Message Handling # ------------------------------------------------------------------------ route[14] { if (!lookup("location")) { sl_send_reply("404", "User Not Found"); exit; }; route(1); } -- View this message in context: http://www.nabble.com/openser-1.2.1-errors-in-the-log-tp19007938p19007938.h… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

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[Kamailio-Users] openser 1.2.1 errors in the log - tcp blocking
by toly 22 Aug '08

22 Aug '08

Greetings, This is openser 1.2.1 compiled with TLS, TLS is not turned on. This is part of the config file: listen=xxx.xxx.xxx.xxx # This is where OpenSER installed port=5070 children=25 tcp_connect_timeout=2 tcp_max_connections=8192 reply_to_via = no sip_warning = yes check_via = no dns=no rev_dns=no disable_dns_blacklist=true here's the problem: I have 25 users having each other as a contact. Softphone is Bria, transport is tcp here comes subscribe: 08/21 15:07:46 8029 debug <loose.c: 115> find_first_route: No Route headers found 08/21 15:07:46 8029 debug <loose.c: 883> loose_route: There is no Route HF 08/21 15:07:46 8029 notice <xlog.c: 148> TRACKING [18] - M=SUBSCRIBE RURI=sip:dmarkiew@tamtam.com F=sip:sgarratt@tamtam.com T=sip:dmarkiew@tamtam.com IP=333.64.100 .15 ID=YTBmOTJhNGI5M2RmZjYyNmM2NGFhMmI1NmRjMmY4ZWE. 08/21 15:07:46 8029 debug <dset.c: 445> rewrite_uri: Rewriting Request-URI with 'sip:dmarkiew@76.222.53.112:53363;transport=TCP;rinstance=575ebcf03ae72b26' 08/21 15:07:46 8029 debug <dset.c: 285> ******* setting for branch 0 flags C0 08/21 15:07:46 8029 debug <dset.c: 285> ******* setting for branch 1 flags C0 08/21 15:07:46 8029 notice <xlog.c: 148> TRACKING [1] - M=SUBSCRIBE RURI=sip:dmarkiew@76.222.53.112:53363;transport=TCP;rinstance=575ebcf03ae72b26 D-URI=sip:76.222.53.11 2:53363;transport=TCP F=sip:sgarratt@tamtam.com T=sip:dmarkiew@tamtam.com IP=333.64.100.15 ID=333.64.100.15 08/21 15:07:46 8029 notice <xlog.c: 148> TRACKING [1] USING NATed reply - M=SUBSCRIBE RURI=sip:dmarkiew@76.222.53.112:53363;transport=TCP;rinstance=575ebcf03ae72b26 D-UR I=sip:76.222.53.112:53363;transport=TCP F=sip:sgarratt@tamtam.com T=sip:dmarkiew@tamtam.com IP=333.64.100.15 ID=333.64.100.15 08/21 15:07:46 8029 debug <t_lookup.c: 1037> DEBUG: t_newtran: T on entrance=0xffffffff 08/21 15:07:46 8029 debug <parser/msg_parser.c: 310> parse_headers: flags=ffffffffffffffff 08/21 15:07:46 8029 debug <parser/msg_parser.c: 310> parse_headers: flags=78 08/21 15:07:46 8029 debug <t_lookup.c: 407> t_lookup_request: start searching: hash=41567, isACK=0 and later I get this: 08/21 15:07:48 8029 err <tcp_main.c: 291> ERROR: tcp_blocking_connect: timeout 2 s elapsed from 2 s 08/21 15:07:48 8029 err <tcp_main.c: 459> ERROR: tcpconn_connect: tcp_blocking_connect failed 08/21 15:07:48 8029 err <tcp_main.c: 721> ERROR: tcp_send: connect failed 08/21 15:07:48 8029 err <../../forward.h: 105> msg_send: ERROR: tcp_send failed 08/21 15:07:48 8029 err <t_fwd.c: 767> ERROR:tm:t_forward_nonack: sending request failed 08/21 15:07:48 8029 debug <tcp_main.c: 718> tcp_send: no open tcp connection found, opening new one And I get a lot of it. Eventually I may get SO_ERROR(111) and then I have to restart proxy. I can post entire config script if anybody had/resolved/willing to take a look at the problem. Thanks a lot, Toly -- View this message in context: http://www.nabble.com/openser-1.2.1-errors-in-the-log---tcp-blocking-tp1909… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

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[Kamailio-Users] Res: Res: Help with SerMyAdmin
by Flavio Goncalves 21 Aug '08

21 Aug '08

Hi Edson, Please check the tomcat logs. They can be found at /usr/local/tomcat6/logs. When tomcat6 don't start, usually is a typo in the configuration files. Best regards, Flavio E. Goncalves ----- Mensagem original ---- De: Edson - Listas <4lists(a)gmail.com> Para: Flavio Goncalves <flavio(a)asteriskguide.com> Cc: Asim Riaz <ariaz78(a)gmail.com>; users <users(a)lists.kamailio.org> Enviadas: Quinta-feira, 21 de Agosto de 2008 14:05:27 Assunto: Re: [Kamailio-Users] Res: Help with SerMyAdmin Yesterday, we faced a similar/related problem, but because Tomcat process was killed. When we try to run it again, SerMyAdmin refuse to start, altought all configurations and DB definitions where there... Any ideas on how to restore it? Edson. Flavio Goncalves escreveu: > Hi Assim, > > I'm glad it is working. SerMyAdmin creates the database fields during > the first run, that's why it worked after restarting the tomcat. The sql > file creates the required users. I'm working to have a debian package > for SerMyAdmin soon. > > Regards, > > Flavio E. Goncalves > > ----- Mensagem original ---- > De: Asim Riaz <ariaz78(a)gmail.com> > Para: users <users(a)lists.kamailio.org> > Enviadas: Quinta-feira, 21 de Agosto de 2008 12:02:50 > Assunto: Re: [Kamailio-Users] Help with SerMyAdmin > > Its working thanks for the help. > I removed my existing serMyAdmin directory from tomcat, restarted the > tomcat and run the script mysql -u root -p openser <openser.sql > > regards > Asim Riaz > > On Thu, Aug 21, 2008 at 3:46 PM, Nuno Marques <yangsengopenser(a)gmail.com > <mailto:yangsengopenser@gmail.com>> wrote: > > How did you add the user? With openserctl or with sermyadmin? > Note that you have to use the same login/pass that are on the sql > script - admin@setup for login and secret for pass. > > > 2008/8/21 Asim Riaz <ariaz78(a)gmail.com <mailto:ariaz78@gmail.com>> > > When I am adding a user i am getting > java.lang.NullPointerException; > > also I cant login using admin username. > > On Thu, Aug 21, 2008 at 3:06 PM, Nuno Marques > <yangsengopenser(a)gmail.com <mailto:yangsengopenser@gmail.com>> > wrote: > > Hi again, > > I was looking at the database schema of openser and i > noticed that the subscriber table only has 12 columns till rpid. > Looking at the openser.sql script, it has more than 12 > columns (including the 'version' column...). > What is wrong? Should SerMyAdmin add this columns to the > subscriber table? > Is it lacking me something in the database subscribet table? > Any help with this? > > Thanks in advance, > > Nuno > > 2008/8/20 muhammad akl <muhammad.akl(a)gmail.com > <mailto:muhammad.akl@gmail.com>> > > > > Hi Nuno > > i faced some problems your already facing and succeeded > to solve them so i'd like to as you > > did you install SERweb tables ? > > if though the problems come from there especially the > phplib_id field a duplication happened between it and > the "id" filed so you have to remove SERweb tables and > all thing will go smoothly without any errors > > p.s i recommend you to remove the whole database e.g > openser and install it again using openser_mysql.sh script > > e.g /path/to/openser_mysql.sh create > > press yes for presence and that stuff but for SERweb no > > try and tell me what you get > > BR > > Muhammad > > > > _______________________________________________ > Users mailing list > Users(a)lists.kamailio.org <mailto:Users@lists.kamailio.org> > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > Users mailing list > Users(a)lists.kamailio.org <mailto:Users@lists.kamailio.org> > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users

1 0

[Kamailio-Users] Res: Help with SerMyAdmin
by Flavio Goncalves 21 Aug '08

21 Aug '08

Hi Assim, I'm glad it is working. SerMyAdmin creates the database fields during the first run, that's why it worked after restarting the tomcat. The sql file creates the required users. I'm working to have a debian package for SerMyAdmin soon. Regards, Flavio E. Goncalves ----- Mensagem original ---- De: Asim Riaz <ariaz78(a)gmail.com> Para: users <users(a)lists.kamailio.org> Enviadas: Quinta-feira, 21 de Agosto de 2008 12:02:50 Assunto: Re: [Kamailio-Users] Help with SerMyAdmin Its working thanks for the help. I removed my existing serMyAdmin directory from tomcat, restarted the tomcat and run the script mysql -u root -p openser <openser.sql regards Asim Riaz On Thu, Aug 21, 2008 at 3:46 PM, Nuno Marques <yangsengopenser(a)gmail.com> wrote: How did you add the user? With openserctl or with sermyadmin? Note that you have to use the same login/pass that are on the sql script - admin@setup for login and secret for pass. 2008/8/21 Asim Riaz <ariaz78(a)gmail.com> When I am adding a user i am getting java.lang.NullPointerException; also I cant login using admin username. On Thu, Aug 21, 2008 at 3:06 PM, Nuno Marques <yangsengopenser(a)gmail.com> wrote: Hi again, I was looking at the database schema of openser and i noticed that the subscriber table only has 12 columns till rpid. Looking at the openser.sql script, it has more than 12 columns (including the 'version' column...). What is wrong? Should SerMyAdmin add this columns to the subscriber table? Is it lacking me something in the database subscribet table? Any help with this? Thanks in advance, Nuno 2008/8/20 muhammad akl <muhammad.akl(a)gmail.com> Hi Nuno i faced some problems your already facing and succeeded to solve them so i'd like to as you did you install SERweb tables ? if though the problems come from there especially the phplib_id field a duplication happened between it and the "id" filed so you have to remove SERweb tables and all thing will go smoothly without any errors p.s i recommend you to remove the whole database e.g openser and install it again using openser_mysql.sh script e.g /path/to/openser_mysql.sh create press yes for presence and that stuff but for SERweb no try and tell me what you get BR Muhammad _______________________________________________ Users mailing list Users(a)lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users(a)lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users

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Re: [Kamailio-Users] Help with SerMyAdmin
by Asim Riaz 21 Aug '08

21 Aug '08

Its working thanks for the help. I removed my existing serMyAdmin directory from tomcat, restarted the tomcat and run the script mysql -u root -p openser <openser.sql regards Asim Riaz On Thu, Aug 21, 2008 at 3:46 PM, Nuno Marques <yangsengopenser(a)gmail.com>wrote: > How did you add the user? With openserctl or with sermyadmin? > Note that you have to use the same login/pass that are on the sql script - > admin@setup for login and secret for pass. > > > 2008/8/21 Asim Riaz <ariaz78(a)gmail.com> > >> When I am adding a user i am getting java.lang.NullPointerException; >> >> also I cant login using admin username. >> >> On Thu, Aug 21, 2008 at 3:06 PM, Nuno Marques <yangsengopenser(a)gmail.com>wrote: >> >>> Hi again, >>> >>> I was looking at the database schema of openser and i noticed that the >>> subscriber table only has 12 columns till rpid. >>> Looking at the openser.sql script, it has more than 12 columns >>> (including the 'version' column...). >>> What is wrong? Should SerMyAdmin add this columns to the subscriber >>> table? >>> Is it lacking me something in the database subscribet table? >>> Any help with this? >>> >>> Thanks in advance, >>> >>> Nuno >>> >>> 2008/8/20 muhammad akl <muhammad.akl(a)gmail.com> >>> >>> >>>> >>>> Hi Nuno >>>> >>>> i faced some problems your already facing and succeeded to solve them so >>>> i'd like to as you >>>> >>>> did you install SERweb tables ? >>>> >>>> if though the problems come from there especially the phplib_id field a >>>> duplication happened between it and the "id" filed so you have to remove >>>> SERweb tables and all thing will go smoothly without any errors >>>> >>>> p.s i recommend you to remove the whole database e.g openser and install >>>> it again using openser_mysql.sh script >>>> >>>> e.g /path/to/openser_mysql.sh create >>>> >>>> press yes for presence and that stuff but for SERweb no >>>> >>>> try and tell me what you get >>>> >>>> BR >>>> >>>> Muhammad >>>> >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users(a)lists.kamailio.org >>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >>> >>> >> >> _______________________________________________ >> Users mailing list >> Users(a)lists.kamailio.org >> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >> >> >

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[Kamailio-Users] Help with SerMyAdmin
by Nuno Marques 21 Aug '08

21 Aug '08

Hi, I´m trying to install SerMyAdmin in my Openser 1.2.2 machine. I follow the steps that were on www.sermyadmin.org. I accomplished the installation task, but now when i try to do something on the DB (for example: create a new user) it gives me a lot of errors. My experience with Grails is null so i don't know were to start. Below i'll show the error messages. If anyone could give me an hand i'll appreciate. Thanks in advance, Nuno *Message:* could not insert: [User]; nested exception is org.hibernate.exception.ConstraintViolationException: could not insert: [User] *Caused by:* org.springframework.dao.DataIntegrityViolationException: could not insert: [User]; nested exception is org.hibernate.exception.ConstraintViolationException: could not insert: [User] *Class:* UserController *At Line:* [147] *Code Snippet:* Stack org.codehaus.groovy.runtime.InvokerInvocationException: org.springframework.dao.DataIntegrityViolationException: could not insert: [User]; nested exception is org.hibernate.exception.ConstraintViolationException: could not insert: [User] at org.apache.catalina.core.ApplicationFilterChain.internalDoFilter(ApplicationFilterChain.java:290) at org.apache.catalina.core.ApplicationFilterChain.doFilter(ApplicationFilterChain.java:206) at org.apache.catalina.core.ApplicationDispatcher.invoke(ApplicationDispatcher.java:630) at org.apache.catalina.core.ApplicationDispatcher.processRequest(ApplicationDispatcher.java:436) at org.apache.catalina.core.ApplicationDispatcher.doForward(ApplicationDispatcher.java:374) at org.apache.catalina.core.ApplicationDispatcher.forward(ApplicationDispatcher.java:302) at org.apache.catalina.core.ApplicationFilterChain.internalDoFilter(ApplicationFilterChain.java:235) at org.apache.catalina.core.ApplicationFilterChain.doFilter(ApplicationFilterChain.java:206) at org.apache.catalina.core.ApplicationFilterChain.internalDoFilter(ApplicationFilterChain.java:235) at org.apache.catalina.core.ApplicationFilterChain.doFilter(ApplicationFilterChain.java:206) at org.apache.catalina.core.ApplicationFilterChain.internalDoFilter(ApplicationFilterChain.java:235) at org.apache.catalina.core.ApplicationFilterChain.doFilter(ApplicationFilterChain.java:206) at org.apache.catalina.core.ApplicationFilterChain.internalDoFilter(ApplicationFilterChain.java:235) at org.apache.catalina.core.ApplicationFilterChain.doFilter(ApplicationFilterChain.java:206) at org.jsecurity.web.servlet.WebInterceptorFilter.doFilterInternal(WebInterceptorFilter.java:106) at org.jsecurity.web.servlet.SecurityContextFilter.doFilterInternal(SecurityContextFilter.java:93) at org.jsecurity.web.servlet.OncePerRequestFilter.doFilter(OncePerRequestFilter.java:106) at org.apache.catalina.core.ApplicationFilterChain.internalDoFilter(ApplicationFilterChain.java:235) at org.apache.catalina.core.ApplicationFilterChain.doFilter(ApplicationFilterChain.java:206) at org.apache.catalina.core.StandardWrapperValve.invoke(StandardWrapperValve.java:233) at org.apache.catalina.core.StandardContextValve.invoke(StandardContextValve.java:175) at org.apache.catalina.core.StandardHostValve.invoke(StandardHostValve.java:128) at org.apache.catalina.valves.ErrorReportValve.invoke(ErrorReportValve.java:102) at org.apache.catalina.core.StandardEngineValve.invoke(StandardEngineValve.java:109) at org.apache.catalina.connector.CoyoteAdapter.service(CoyoteAdapter.java:286) at org.apache.coyote.http11.Http11Processor.process(Http11Processor.java:844) at org.apache.coyote.http11.Http11Protocol$Http11ConnectionHandler.process(Http11Protocol.java:583) at org.apache.tomcat.util.net.JIoEndpoint$Worker.run(JIoEndpoint.java:447) at java.lang.Thread.run(Thread.java:595) Caused by: org.springframework.dao.DataIntegrityViolationException: could not insert: [User]; nested exception is org.hibernate.exception.ConstraintViolationException: could not insert: [User] at UserController$_closure8.doCall(UserController.groovy:147) at UserController$_closure8.doCall(UserController.groovy) ... 29 more Caused by: org.hibernate.exception.ConstraintViolationException: could not insert: [User] at $Proxy3.saveOrUpdate(Unknown Source) ... 31 more Caused by: com.mysql.jdbc.exceptions.MySQLIntegrityConstraintViolationException: Duplicate entry '4000-192.168.137.1' for key 2 at com.mysql.jdbc.SQLError.createSQLException(SQLError.java:1011) at com.mysql.jdbc.SQLError.createSQLException(SQLError.java:956) at com.mysql.jdbc.MysqlIO.checkErrorPacket(MysqlIO.java:3491) at com.mysql.jdbc.MysqlIO.checkErrorPacket(MysqlIO.java:3423) at com.mysql.jdbc.MysqlIO.sendCommand(MysqlIO.java:1936) at com.mysql.jdbc.MysqlIO.sqlQueryDirect(MysqlIO.java:2060) at com.mysql.jdbc.ConnectionImpl.execSQL(ConnectionImpl.java:2542) at com.mysql.jdbc.PreparedStatement.executeInternal(PreparedStatement.java:1734) at com.mysql.jdbc.PreparedStatement.executeUpdate(PreparedStatement.java:2019) at com.mysql.jdbc.PreparedStatement.executeUpdate(PreparedStatement.java:1937) at com.mysql.jdbc.PreparedStatement.executeUpdate(PreparedStatement.java:1922) at org.apache.tomcat.dbcp.dbcp.DelegatingPreparedStatement.executeUpdate(DelegatingPreparedStatement.java:102) ... 32 more

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[Kamailio-Users] Fwd: Help with SerMyAdmin
by Asim Riaz 21 Aug '08

21 Aug '08

---------- Forwarded message ---------- From: Asim Riaz <ariaz78(a)gmail.com> Date: Thu, Aug 21, 2008 at 2:49 PM Subject: Re: [Kamailio-Users] Help with SerMyAdmin To: Nuno Marques <yangsengopenser(a)gmail.com> my issue is same as Nano tried this command mysql -u root -p openser <openser.sql but still got ERROR 1054 (42S22) at line 1: Unknown column 'version' in 'field list' Regards Asim On Thu, Aug 21, 2008 at 12:26 PM, Nuno Marques <yangsengopenser(a)gmail.com>wrote: > Hi Muhammad, > > I droped the database and then created a new database as you told me... > Now i have to create all users again and an admin for sermyadmin... > But when i tried to run the mysql commands that are on > www.sermyadmin.org/openser/openser.sql they gave me an error: unknown > column 'version' in 'field list'. > Have any idea of what is going on? > By the way... do you know what happened to www.sermyadmin.org - it's seem > like it has been hacked or something... > > Many tahnks, > > Nuno > > > 2008/8/20 muhammad akl <muhammad.akl(a)gmail.com> > > >> >> Hi Nuno >> >> i faced some problems your already facing and succeeded to solve them so >> i'd like to as you >> >> did you install SERweb tables ? >> >> if though the problems come from there especially the phplib_id field a >> duplication happened between it and the "id" filed so you have to remove >> SERweb tables and all thing will go smoothly without any errors >> >> p.s i recommend you to remove the whole database e.g openser and install >> it again using openser_mysql.sh script >> >> e.g /path/to/openser_mysql.sh create >> >> press yes for presence and that stuff but for SERweb no >> >> try and tell me what you get >> >> BR >> >> Muhammad >> > > > _______________________________________________ > Users mailing list > Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > >

2 1

[Kamailio-Users] Change project name in Surceforge?
by Iñaki Baz Castillo 21 Aug '08

21 Aug '08

Hi, is not time to change the project name in SourceForge? XD It appears still as "OpenSER SIP Server". -- Iñaki Baz Castillo ibc(a)in.ilimit.es

2 1

[Kamailio-Users] SIP Session timeouts
by Jason Penton 21 Aug '08

21 Aug '08

Hi All, I have a question about the best solution to ensure that SIP calls are terminated when the 2 parties drop off the network (wireless). Is there a std way of doing this (maybe using sst module) or should this be implemented using some specialised code? If not supported in a std manner in Kamailio, i thought about creating a child process within the DLG module that continuously checks all dialogs and the connected parties and pings them for connectivity. Should connectivity of any participant drop for more than a specific unit of time then the call is focefully torn down by openser. Any ideas? Cheers Jason

2 2

[Kamailio-Users] RATELIMIT link on modules docs page?
by Alex Balashov 21 Aug '08

21 Aug '08

Why does the RATELIMIT link on http://www.kamailio.org/docs/modules/1.4.x/ go to the documentation for the Perl module? -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599

3 2

[Kamailio-Users] Permissions module error !
by Nuno Marques 21 Aug '08

21 Aug '08

Hi, I tried to install on my Openser 1.2.2 the permissions module so i can use an asterisk gateway. I populated the trusted table with information about the gateway. When i tried to run OpenSer it gave me some errors. Can anyone tell me what could be the problem? Below are the erros and the data on trusted table. Thanks in advance, Nuno 0(3824) init_tcp: using epoll_lt as the io watch method (auto detected) 0(0) INFO: statistics manager successfully initialized 0(0) StateLess module - initializing 0(0) TM - initializing... 0(0) Maxfwd module- initializing 0(0) INFO:ul_init_locks: locks array size 512 0(0) TextOPS - initializing labvoipua:/etc/openser# 0(0) permissions - initializing 0(0) Default allow file (//etc/openser/permissions.allow) not found => empty rule set 0(0) Default deny file (//etc/openser/permissions.deny) not found => empty rule set 0(0) hash_table_insert(): Unknown protocol 0(0) ERROR: permissions: trusted_reload(): Hash table problem 0(0) init_trusted(): Reload of trusted table failed 0(0) Error while initializing allow_trusted function 0(0) init_mod(): Error while initializing module permissions ERROR: error while initializing modules 0(0) INFO:mi_fifo:mi_destroy:memory for the child's mi_fifo_pid was not allocated -> nothing to destroy TRUSTED TABLE +----+----------------+-------+--------------+------+---------+ | id | src_ip | proto | from_pattern | tag | version | +----+----------------+-------+--------------+------+---------+ | 1 | 192.168.137.20 | Any | ^sip:.*$ | NULL | 3 | +----+----------------+-------+--------------+------+---------+

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[Kamailio-Users] Speed dial
by salman farci 21 Aug '08

21 Aug '08

Hi, I have openser 1.2.2 installed with mysql as backend. I was able to run a simple call service after registering users in location table. Now that I'm trying implement a speed_dial, it returns "404 call not found" error at the Xlite panel after dialing a short dial number as per speed_dial table entry. But if I short dial my own short dial number it dials back to me(working) ! What might b the problem ? script file : ........... if (uri=~"sip:[0-9]{2}@.*") sd_lookup("speed_dial"); ............ speed_dial table entry : +----+------------+----------+----------------+--------------+----------------+-----------+----------+-----------------+ | id | username | domain | sd_username | sd_domain | new_uri | fname | lname | description | +----+------------+----------+----------------+--------------+----------------+-----------+----------+-----------------+ | 5 | 103 | | 33 | 10.1.2.214 | 103(a)10.1.2.214 | | | 4 | 102 | | 22 | 10.1.2.214 | 102(a)10.1.2.214 | | | 6 | 101 | | 11 | 10.1.2.214 | 101(a)10.1.2.214 | | | 9 | 105 | | 55 | 10.1.2.214 | 105(a)10.1.2.214 | | +----+------------+----------+----------------+--------------+----------------+----------+----------+-----------------+ can anyone clarify ? Thanx regards Farci

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[Kamailio-Users] Second Logo
by Darren Sessions 20 Aug '08

20 Aug '08

Good work. I tinkered a bit and came up with something very similar. - Darren _____________________________ dmsessions(a)gmail.com http://www.darrensessions.com _____________________________ On Aug 20, 2008, at 4:52 PM, Ricky Gutierrez wrote: > logo with basic colors > > Regards > > rickygm > > ----- Original Message ---- > From: Johansson Olle E <oej(a)edvina.net> > To: Ricky Gutierrez <xserverlinux(a)yahoo.com> > Cc: devel(a)lists.kamailio.org > Sent: Wednesday, August 20, 2008 4:55:21 AM > Subject: Re: [Kamailio-Devel] logo > > Ricky, > I really like your logo. But I also see issue with using it in pure > black/white situations - no greyshades. > > Can you try to create such a version of it as well? > > This is often needed when producing t-shirts and other stuff, where an > outline is used. > > Thanks. > /O > > <kamailio-new.png>_______________________________________________ > Devel mailing list > Devel(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/devel

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[Kamailio-Users] Commercial support?
by Robert McGilvray 20 Aug '08

20 Aug '08

With the departure of voice system from the project are there any commercial entities associated with the development team that offer support for Kamailio? Regards, Bob -------------------------------------------------------------------------- This email with all information contained herein or attached hereto may contain confidential and/or privileged information intended for the addressee(s) only. If you have received this email in error, please contact the sender and immediately delete this email in its entirety and any attachments thereto. (W1)

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[Kamailio-Users] How to test and trace the RLS messages?
by mahesh.peddi＠info-spectrum.com 20 Aug '08

20 Aug '08

Hi All, I am working on openser 1.3.2 version with RLS as "integrated_xcap_server". Its running without erros. But my query is how to test, And please send me call flow between IMS components and RLS server. Regards, Mahesh Peddi Infospectrum India Pvt. Ltd. Cell: +91 9765775176 IP-Phone Ext. - 764

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Re: [Serusers] Content-Type: application/isup read encoded message
by Jiri Kuthan 19 Aug '08

19 Aug '08

I think the best learning experience is OOB http://cvs.berlios.de/cgi-bin/viewcvs.cgi/ser/sip_router/etc/ser-oob.cfg?re… it is not really simple, on the contrary -- it is quite complete, but for that it gives you all kinds of examples that can be probably useful to you. -jiri michael(a)teldelnort.com wrote: > Hello jiri, > > Ok, > > Do yo have an idea of a simple example that will help me familiarize the > usage in ser.cfg? > > -michael > > -----Original Message----- > *From:* Jiri Kuthan [mailto:jiri@iptel.org] > *Sent:* Tuesday, August 19, 2008 03:44 PM > *To:* michael(a)teldelnort.com > *Subject:* Re: [Serusers] Content-Type: application/isup read > encoded message > > I don't think so for this particular one -- just try to use it as > other select operators. -jirimichael(a)teldelnort.com wrote:> Hello > Jiri,> > Thanks for the response.> > Is there any documentation on > how to use select operator with @context?> > > I am trying to filter > a search on the context for "II Digits:"> > -michael> > > > -----Original Message-----> *From:* Jiri Kuthan > [mailto:jiri@iptel.org]> *Sent:* Tuesday, August 19, 2008 02:09 AM> > *To:* michael(a)teldelnort.com> *Cc:* serusers(a)iptel.org> *Subject:* > Re: [Serusers] Content-Type: application/isup read> encoded message> > > I think the select operator, @content, should get you access to > the> body. There is however noparsing of SDP to my knowledge > (unless> someone corrects me, I remember it has been> > frequentlymentioned.)-jirimichael(a)teldelnort.com wrote:> Hello,> >> > How can I evaluate the Content-Type: multipart/mixed;boundary= with> > SER?> > Content-Type: application/isup; base=ansi92; version=ansi> > >> Is there a module that allows SER read the contents of the > encoded> message?> I am try to get the > > "Originating Line > Information> ----> II Digits:"> > from the message.> > -michael> > > >> > ------------------------------------------------------------------------>> > > _______________________________________________> Serusers > mailing> list> Serusers(a)lists.iptel.org>> > http://lists.iptel.org/mailman/listinfo/serusers

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[Kamailio-Users] Ratelimit outgoing SIP INVITE messages after dispatcher
by sergejf 19 Aug '08

19 Aug '08

Hello, We are connected to several VoIP carriers that impose a calls per second (CPS) rate limit on their SIP trunks. For example, on TRUNK1 we can only make call attempts at 10 CPS. Our calls are originally made by a dialer that does not support rate limiting, hence it always blasts calls as fast as it can. Upon receiving these SIP INVITE messages we determine what carrier to use and perform a mapping to a dispatcher group. Call attempts are then load-balanced between all SIP trunks from the chosen carrier. My question is whether it is possible to use the ratelimit module in conjunction with the dispatcher (or carrierroute) to ensure we don't overload the carrier SIP trunks? If this is possible, then I wonder how to configure ratelimit to support a 10 CPS call attempts rate. Thanks in advance for any advice on the topic! Serge -- View this message in context: http://www.nabble.com/Ratelimit-outgoing-SIP-INVITE-messages-after-dispatch… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

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[Serusers] Content-Type: application/isup read encoded message
by michael＠teldelnort.com 19 Aug '08

19 Aug '08

Hello, How can I evaluate the Content-Type: multipart/mixed;boundary= with SER? Content-Type: application/isup; base=ansi92; version=ansi Is there a module that allows SER read the contents of the encoded message? I am try to get the "Originating Line Information ---- II Digits:" from the message. -michael

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[Kamailio-Users] About Asterisk2Billing
by Ali Jawad 19 Aug '08

19 Aug '08

Hi All Was anyone able to get the accounting part of asterisk2billing to work with Openser ? This looks like a pretty mature solution to me. Thanks With Regards Ali Jawad System Administrator Splendor Telecom (www.splendor.net) Beirut, Lebanon Phone: +961 1 373725 Fax: + 961 1 375554

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[Kamailio-Users] Accounting for OPenser
by Ali Jawad 19 Aug '08

19 Aug '08

Hi I am looking for a accounting solution, that I can use with Openser without involving Asterisk. Has anyone tested something similar ? I need the solution to work for my prepaid customers, for example he has got 10$ once the credit is used. Deny any more call.s With Regards Ali Jawad System Administrator

3 2

[Kamailio-Users] SIPp dispatcher load testing failure with too many hops - why?
by sergejf 18 Aug '08

18 Aug '08

Hello, I'm attempting to load test a simple dispatcher script in OpenSER 1.3.x using SIPp (built-in UAC and UAS scenarios). My dispatcher.list only has one address 8.XX.XX.12 (a SIPp instance, UAS). SIPp on 8.XX.XX.10 sends the INVITE messages to 63.XXX.XXX.110. Everything goes well until the UAC sends ACK and BYE in quick sequence. OpenSER tries to t_relay() it to itself, which results in a "too many hops" error: 2008-08-18 11:16:56:708 1219072616.708649: Aborting call on unexpected message for Call-Id '1-3548(a)8.XX.XX.10': while expecting '200' (index 8), received 'SIP/2.0 483 To Many Hops Via: SIP/2.0/UDP 8.XX.XX.10:5061;branch=z9hG4bK-3548-1-7 From: sipp <sip:sipp@8.XX.XX.10:5061>;tag=3548SIPpTag001 To: sut <sip:service@63.XXX.XXX.110:5060>;tag=2148SIPpTag011 Call-ID: 1-3548(a)8.XX.XX.10 CSeq: 2 BYE Server: OpenSER (1.3.2-notls (x86_64/linux)) Content-Length: 0 I have tried modifying the SIPp UAC script to re-use the Contact: header from the SIP 200 OK message in the R-URI of the SIP ACK and SIP BYE, to no avail. I have loaded the TM module but I suppose that since the SIP 200 OK is the final message of the transaction, SIP ACK and SIP BYE won't automatically be relayed to 8.XX.XX.12. Is this an OpenSER issue or a SIPp scripting issue? Any advice would be appreciated. Please find below some relevant lines from my openser.cfg: route{ xlog("TRACE:ROUTE: src($si:$sp) dst($Ri:$Rp) msg($mb)\n"); # initial checkings if ( !mf_process_maxfwd_header("10") ) { xlog("SCRIPT:ERROR: $rm (from $si:$sp) too many hops\n"); sl_send_reply("483","To Many Hops"); exit; }; if (method==CANCEL) { if (t_check_trans()) t_relay(); exit; } # routing if (has_totag()) { xlog("SCRIPT0:INFO: $rm RURI=[$ru] - routing to dst-uri [$du] cnt [$avp(i:273)] dst set [$avp(i:271)]\n"); loose_route(); if (method=="INVITE") record_route(); route(1); exit; } The SIP ACK and SIP BYE follow the route defined in the above if {} statement. The SIP INVITE follows the dispatcher route shown below: record_route(); # perform load balancing # set algorithm (4 = round robin) $avp(alg) = 4; # set the group $avp(grp) = 2; if (method=="INVITE") xlog("CALL_START: RURI=[$ru] group=[$avp(grp)] callid=$ci\n"); # do balancing if(!ds_select_dst("$avp(grp)", "$avp(alg)")) { xlog("CALL_DROP:INTERNAL: no destinations for [$ru] group=[$avp(grp)], callid=$ci\n"); sl_send_reply("404", "no dst"); exit; } if(avp_check("$avp(i:273)", "eq/i:1")) { if(!avp_pushto("$ruri","$avp(i:271)")) { xlog("CALL_DROP:INTERNAL: cannot push to ruri [$avp(i:271)], group=[$avp(grp)], callid=$ci\n"); sl_send_reply("500", "cannot get dst"); exit; } } else { # redundancy t_on_failure("1"); } xlog("SCRIPT:INFO: $rm RURI=[$ru] - routing to dst-uri [$du] cnt [$avp(i:273)] dst set [$avp(i:271)]\n"); # do forward route(1); exit; -- View this message in context: http://www.nabble.com/SIPp-dispatcher-load-testing-failure-with-too-many-ho… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

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[Kamailio-Users] poll: project name
by Daniel-Constantin Mierla 18 Aug '08

18 Aug '08

Hello everybody, some people expressed inconveniences regarding the new name, so we would like to get your opinion about, therefore this is a poll that will get us to a decision. Here is the question: Do you like the name "Kamailio"? 1) Yes, I love it. Please keep it. 2) It is OK. 3) I do not mind the name at all 4) I do not like it 5) I hate the name and we should change it There is no need of explanations, as it is up to personal taste and decision. Just vote by option. Cheers, Daniel PS. for those interested... As short history about renaming story. On the 28th of March the board was informed about issues with SER part of project name. You can easily find who has the trademarks. On the 7th of April the board was informed about an agreement for the deadline as of September 30, 2008 to use old name. From there on, all old releases, documents, etc. can be referred by the old name, but nothing new. Other few discussions about what should/could be kept/changed followed and on 12th of May started the discussion about the new name with two main direction, something similar, to remind about old project or just something completely new, to be used as brand name. Couple of proposals and discussions regarding possible names followed up to 16th of July when was the deadline and the start of vote. Majority was reached on 17th but it was required to have opinions from all members of the board, last came from Bogdan, on the 28th of July. All discussions about migration were on the board mailing list. No change were done to openser.org configuration, just the new domain with kamailio in it was pointed to same server. Every step done was reported to board, following a schedule discussed there. Another discussion was about making the renaming and 1.4.0 release in same day. Being practically impossible to do all the work in one day, it was decided to do first the announcement about renaming. That happened on 28th when all agreed with the new name. As you can see, there was no 5 months of renaming duration. There was no change in openser.org DNS configuration undertaken with/without the knowledge of all board members. The new name domains were reserved when majority was achieved, in any case; dns entries for new name were pointed to old server shortly after booking the domains. -- Daniel-Constantin Mierla http://www.asipto.com

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[Serusers] SIP and IP multicast
by Steven C. Blair 18 Aug '08

18 Aug '08

Has anyone had any experience using SIP via SER to setup an IP multicast session with a group of phones?

2 1

[Kamailio-Users] Looking for a VOIP Engineer
by Jonathan K. Creasy 18 Aug '08

18 Aug '08

Voxitas Inc. is a national VOIP provider based in St. Louis, MO. and we are currently looking for qualified applicants to fill Tier Three support positions and a VOIP Engineer position. The reason I am posting to this list is because we are looking to fill the VOIP Engineer position with someone who has OpenSER experience and is comfortable configuring and maintaining these systems. For details please contact our VP of Operations Zack Sargent. Zack Sargent VP Operations and Engineering Voxitas, Inc. 210 N. Tucker, St. Louis, MO. 63101 314-266-4000 x108 zsargent(a)voxitas.com Thank you! -Jonathan

1 0

[Kamailio-Users] PortaBilling
by Ali Jawad 18 Aug '08

18 Aug '08

Hi All Has anyone tried portabilling with Openser ? If so can you please provide a sample of your config file ? THanks With Regards Ali Jawad System Administrator Splendor Telecom (www.splendor.net) Beirut, Lebanon Phone: +961 1 373725 Fax: + 961 1 375554

1 0

[Serusers] SER - Waiting Transaction
by Arun Kumar 18 Aug '08

18 Aug '08

Hi, I'm running SER 0.96 in my production for a long but there is strange problem , when I do serctl moni and looks for transaction it shows waiting transaction in () almost 400+ can some one tell me why there are so many transaction in waiting while I see processing only 10-20 transaction. Thanks arun

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[Kamailio-Users] pick_branch failed (lowest==-1) for code 302--HELP!!
by Krunal Patel 18 Aug '08

18 Aug '08

Hi, Here is the scenario: -----------------------Invite--------------> <----------100 Trying---------------------- <----------302 Moved temporarily-------- -----------------------ACK--------------> Now I would like to route call to contact returned in 302 responce. But I am getting problem: Aug 17 18:10:41 [16667] DBG:core:parse_headers: flags=80 Aug 17 18:10:41 [16667] DBG:tm:t_should_relay_response: T_code=100, new_code=302 Aug 17 18:10:41 [16667] DBG:tm:t_pick_branch: picked branch -1, code 999 Aug 17 18:10:41 [16667] CRITICAL:tm:t_should_relay_response: pick_branch failed (lowest==-1) for code 302 Aug 17 18:10:41 [16667] DBG:tm:relay_reply: branch=0, save=0, relay=-1 Please help me to fix this. Thanking you Krunal Patel

1 1

[Serusers] Gentilini, Paul is out of the office.
by PGentilini＠intellicare.com 17 Aug '08

17 Aug '08

I will be out of the office starting 08/15/2008 and will not return until 08/18/2008. Please contact the CSC at 888-899-4227 if you need immediate assistance.

1 0

[Serusers] Acc module extra patches
by caio 16 Aug '08

16 Aug '08

Hello, Is there patches for acc extra params for ser version 0.9.7? I've found this link http://www.voice-system.ro/docs/acc-headers/ but do not know if is it for v0.9.x Thanks... -- caio

1 0

[Serusers] on Ubuntu Installation & Config
by Ameed Jamous 16 Aug '08

16 Aug '08

Hi all, Am trying to install & Configure SER on ubuntu 8.04. After i intsalled the SER file i can see this in my terminal: Listening on udp: 127.0.0.1 [127.0.0.1]:5060 udp: 192.168.1.104 [192.168.1.104]:5060 tcp: 127.0.0.1 [127.0.0.1]:5060 tcp: 192.168.1.104 [192.168.1.104]:5060 Aliases: tcp: Ubuntu-AM.local:5060 tcp: localhost:5060 udp: Ubuntu-AM.local:5060 udp: localhost:5060 ok now how can i add a user to register to SER ? How can i monitor the logs and the prosses happening? i tried using serctl moni Error opening ser's FIFO /tmp/ser_fifo Make sure you have line fifo=/tmp/ser_fifo in your config Even though i have line fifo=/tmp/ser_fifo in my config. Can anyone help with this? Regards, Ameed -- Best Regards, Ameed Jamous Project Manager/Carriers Relations. Direct Line: +962 777 835819. www.joodtelecomgroup.com

1 0

[Serusers] call forward no answer
by caio 15 Aug '08

15 Aug '08

Hi all, I can't make call forwarding on no answer works on my ser-0.9.7 installation. Here you can see a snippet of log, where the call falls on failure_route and then to route_1 where t_relay is called. Log at: http://rafb.net/p/xwTmn195.html The call is from sip:579951 to sip:579915(who has enabled the feature fwdnoanswer to sip:579916. But I do not know why the INVITE after the append_branch() is still to R-URI: sip:579915@.... After the relay then the call terminates with: Aug 15 16:13:25 [/usr/sbin/ser] SIP Reply (status):_ Aug 15 16:13:25 [/usr/sbin/ser] version: <SIP/2.0>_ Aug 15 16:13:25 [/usr/sbin/ser] status: <487>_ Aug 15 16:13:25 [/usr/sbin/ser] reason: <Request Terminated>_ I follow the example of ser guide.., but do not work.. I would appreciate any comment.. Regards, -- caiogf

1 0

[Kamailio-Users] Question
by Edgar Guadamuz 15 Aug '08

15 Aug '08

Hello, I'm a newbie and I don't know the difference between opensips (where openser.org redirects) and kamailio, If someone can give me a light on this it would be great, thanks,

3 2

[Kamailio-Users] project foundation
by Daniel-Constantin Mierla 15 Aug '08

15 Aug '08

Hello everybody, I think it is the time to move forward with project management structure. During last days there were talks and discussions on the lists, the solution to make the project independent and long term trustable is to create a foundation. We made a draft of purpose and organization structure and we want to get your comments. http://www.kamailio.net/dokuwiki/doku.php/admin:foundation-draft For the statute we are looking at similar cases of foundations around open source projects. Considering the developer population and contributors, I find as good and reliable place to register the foundation to be Germany. There is still long way to go and a lot to discuss: * membership fees * how one can become a full member (associate member can be anyone that is friend of the project and pays the membership fee) * how a member loses the board or full member rights * how the board is supervised o when and how the reports of the board are presented o when and how the general assembly meeting takes place o how the reports are approved by full members * how to change the statute over time ... and perhaps more, but if we start and put some effort we can have it ready in reasonable time, avoiding trust and reliability issues in long term. Let's have debates here and contribute the results to the wiki. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

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[Kamailio-Users] project status and my decisions
by Daniel-Constantin Mierla 15 Aug '08

15 Aug '08

Hello all, following the last events everybody perhaps wonders what is going on. I was surprised by the way everything happens and I am not sure yet how this resulted. Probably a bit of time is needed to clarify relations between developers themselves and community as well. There is not any official statement that a developer quited the support for this project, but I guess the trust between some is at lowest level possible. It is clear that we cannot say anything about a board right now that manages the project. From board discussions summary you may see that discussions were conducted to create a legal entity around the project, a non profit foundation. This might be hard to achieve in this status, but as a personal decision, I am not going in any other management group unless there is a legal framework backing up. Human relations prove to be pretty unpredictable and not conforming with initial statements. What is sure now - openser 1.3.x and 1.4.x will be maintained properly and I am volunteering to take care but hope others will join. It is really a good achievement of a sip server. Decision about future development will be based on community and developers feedback for this project. I am not going to promise anything by myself as I admit one person (to be more specific: myself) cannot take care of the entire code base -- I am just realistic, not willing to sell dreams. I think the people that still believe in the project can come with suggestions and alternatives of future development. What I can tell for sure, I am not going to join a fork of the original openser project. If something happened to make developers not working together, solution to solve nicely could have been found, in the worst, maybe just by splitting teams, but keeping old domain together just to show the new alternatives. "Why not joining a fork in this manner?"> Because I was one of the founders of this one and involved in development and management at top level. If I failed to build something I started, I had all the tools in my hands, why just creating something similar saying the old is crap but this new one will be the best? What is the difference? Why I didn't do it first time? Would I be able to fool someone? Does leaving my ship in secrecy on back door will bring more trust on me? I doubt and perhaps you, too. I am certainly assuming share of the failure in managing this project. I was there, in management, period. There is no single person responsible for mis-management, but it comes just easy for some to blame the others. But I am really confident we built outstanding sip server, all of us contributed with code, documentation, testing to an application that routes million and millions of minutes per day. There were accuses of bad things about project, I will argument in a separate message, this will get too long. So, I am asking: - how do you see a proper management structure for the future - feedback about future involvement, suggestions and alternative for going on Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

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[Serusers] Register to SER with UDP
by Steven Wu 15 Aug '08

15 Aug '08

Hi All, Does SIP server at iptel.org support UDP register? When my UA rigister to iptel.org whith UDP, the 401 response always replied by SER at port 100xx(example 10028). But in my contact of rigister port assigned to 5060. So, my UA can't recive the 401 response, register failure. But use TCP everyting is OK? Can anyone give me help? Best regards, Steven Wu

2 1

[Kamailio-Users] KAMAILIO v1.4.0 is released
by Daniel-Constantin Mierla 15 Aug '08

15 Aug '08

KAMAILIO v1.4.0 is released August 7, 2008 Kamailio is the new name of the SIP server known so far as OpenSER. Due to trademark issues, the OpenSER project had to change the name. Officially, the new name was announced on July 28, 2008. Version 1.4.0 brings many new features since OpenSER v1.3.0 as well as improvements to existing components. An important share of effort was directed to code cleanup and stability. Moreover, new features in core and old modules were added while other 6 new modules were introduced. Download: You can download the tarball of the released sources at http://www.kamailio.net/pub/kamailio/latest/src/ Binary packages for several distributions you can find at http://www.kamailio.net/pub/kamailio/latest/bin/ http://www.kamailio.net/pub/kamailio/latest/packages/ Packages will be uploaded as soon as they are submitted. Documentation: * modules' documentation: http://www.kamailio.net/docs/modules/1.4.x/ * migration guide : http://www.kamailio.net/dokuwiki/doku.php/install:1.3.x-to-1.4.0 Summary of the new stuff in core: * overhauled DB API - better performance and safety, more common functionality integrated in the core * extensive cleanups in database drivers - integrate common functionality into the core for more stability and maintainability * SDP parser - provides an internal API for parsing SDP * fixup functions - provides an internal API for fixing module parameters * new free fixup functions - making safer to use some functions exported by modules in embedded languages such as Perl * extension to module interface - its now possible to use up to 6 parameters in module functions New modules: * db_oracle - ORACLE DB driver - connect to Oracle SQL Server * dialplan - Dialplan management - dialplan regular expression based translations * nat_traversal - NAT traversal helper * peering - RADIUS based peering allowing SIP providers to verify via a broker if source or destination request is from a trusted peer * ratelimit - SIP ttraffic shaping control and server load control based on your rules * userblacklist - User-specific blacklists New in existing modules: * acc module * it is possible to log accounting related output to a different log facility * cfgutils module * new FIFO functions get_config_hash and check_config_hash for config file checks * shared PV exported to configuration script * debugging helpers usable from config script for abort, shm_status and pkg_status * mysql module * enable timeouts on connect, read and write to prevent blocking on errors * database modules * renaming of all database modules, addition of the prefix 'db_', e.g. mysql is now db_mysql * carrierroute module * extension: much more flexibility in routing and database supported failure routing, improved internal structure * dialog module * internal API reworked for better flexibility * direction of the message provided via the internal API * new dialog callback types * new mi command: dlg_list_ctx * dispatcher module * ability to load destination URI flags from database at startup or reload * new algorithm to hash the content of a pseudo-variable * enum module * enum_fquery([...]) replaced by enum_pv_query([...]) * msilo module * notification system refurbished - message body, content-type, from address and contact header can be dynamically specified with pseudo-variables * pdt module * accepts now same destination domain bound to different prefixes * internal structures and logic optimized for memory and performance * lcr module * refactor module functions * pseudo-variable support * mediaproxy module * update to version 2.0 * better performance and scalability as packets are forward in kernel space New testing suite * a testing suite to ensure quality checking and regression tests reports has been developed, included in the repository and used during this development cycle Other important changes * modules' documentation has been migrated to DOCBOOK XML format * doxygen documentation extended and developer guide published * serweb tables have been removed * command line tool to manage dbtext in a SQL fashion These are not all, there is a big change log that gives more details: * http://www.kamailio.net/pub/kamailio/latest/ChangeLog More on http://www.kamailio.net Best regards, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

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[Kamailio-Users] As a board member
by Adrian Georgescu 15 Aug '08

15 Aug '08

Hello, Because of sheer incompetence and abuse conduct of one of the recently adopted board members I decided to put my activities as a management board member of OpenSER/Kamailio project on hold. I will be able to resume my activities after the voluntary leave of Henning Westerholt. For those who do not know who I am, I was involved with OpenSER from Day 0 of its inception. Kind regards, Adrian

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[Serusers] SIP conference.
by Peter Cai 15 Aug '08

15 Aug '08

Hi all! I know H323 support conference but SIP says nothing about it. I want to make some study on this subject but there seems to be little information on Internet. Could some one give me some clues? Is there any RFC draft on this concept?

2 1

[Kamailio-Users] Voicemail implementation...
by Gerard A. Matthew 14 Aug '08

14 Aug '08

Ok, I'm trying to get the voicemail system working, however for some reason my Asterisk box is receiving a duplicated IP Address on the c line from OpenSER....below is the sample sip header. Is there anything in my openSER config file that would cause such a duplication? Rgds, Gerard INVITE sip:u5073@<DOMAIN>:5061 SIP/2.0 Record-Route: <sip:<DOMAIN_IP>;ftag=1d498377;lr=on> Via: SIP/2.0/UDP <DOMAIN_IP>;branch=z9hG4bKaed4.96adae34.1 Via: SIP/2.0/UDP 192.168.1.112:51006;received=<CLIENT_XIP>;branch=z9hG4bK-d8754z-0127430e81514e54-1---d8754z-;rport=51006 Max-Forwards: 69 Contact: <sip:0078@<CLIENT_XIP>:51006;nat=yes;nat=yes> To: "5073"<sip:5073@<DOMAIN>> From: "G Laptop"<sip:0078@<DOMAIN>>;tag=1d498377 Call-ID: NDZiOTUyNWFmYTQ3M2JlMjNiYzdmZjcyMGYyZDljZWE. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: X-Lite release 1100l stamp 47546 Content-Length: 481 P-hint: usrloc applied v=0 o=- 6 2 IN IP4 192.168.1.112 s=CounterPath X-Lite 3.0 *c=IN IP4 <XXX.XX.XXX.XXX><XXX.XX.XXX.XXX>* t=0 0 m=audio 3649036490 RTP/AVP 107 119 100 106 0 105 98 8 3 101 a=alt:1 1 : 4IoIRnGz iYQAZz1E 192.168.1.112 16720 a=fmtp:101 0-15 a=rtpmap:107 BV32/16000 a=rtpmap:119 BV32-FEC/16000 a=rtpmap:100 SPEEX/16000 a=rtpmap:106 SPEEX-FEC/16000 a=rtpmap:105 SPEEX-FEC/8000 a=rtpmap:98 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv a=nortpproxy:yes a=nortpproxy:yes

2 1

[Kamailio-Users] Translate name in RURI to number
by Ali Jawad 14 Aug '08

14 Aug '08

Hi all..my users are all registered with names...I want to forward their calls to my pstn gw..however those only take numerical ANI numbers. So I need to change the sip packet on the Openser server to a number (diff number per user) before forwarding that to the PSTN Gateway ..any ideas ? Would ENUM work ? THanks

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[Kamailio-Users] [OT] Any domain registrar offering SRV, NAPTR and AAAA?
by Iñaki Baz Castillo 14 Aug '08

14 Aug '08

Hi, first of all I'm sorry for the OffTopic. I'm looking for a domain registrar and nameserver that offers me SRV, NAPT and AAAA registers ni their domain control panel. Unfortunatelly in Spain *NO ONE* offers it. Is there any one "outside"? Thanks a lot. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

1 0

[Kamailio-Users] Kamailio & CDRTool 6.5.XX & MediaProxy 2.0
by luzango mfupe 14 Aug '08

14 Aug '08

Hi guys,I read the release note of MediaProxy 2.0 that to be able to have it fully functioning one needs CDRTool version 6.5.X and above i have been searching all over with no avail can someone point where it if found?? by the way i just found the old version 6.4.1 in the author's site wbr, Lu. -- Luzango Mfupe TUUNE MOBILE Tel:0128440528/0123825710 Tshwane-RSA "...Ships are safe in harbor, but they were never meant to stay there......."

1 0

[Kamailio-Users] new developer: Carsten Bock
by Daniel-Constantin Mierla 14 Aug '08

14 Aug '08

Hello, I wish to welcome long time contributor, Carsten Bock, among developers. He is going to be officially co-maintainer of the dispatcher module and help with several other modules such as dialog, uac, ... He has submitted lot of new features in the past to many parts of the code in openser. His id is 'carstenbock'. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 0

[Kamailio-Users] openser 1.2.1 errors in the log
by toly 14 Aug '08

14 Aug '08

Greetings, This is openser 1.2.1 compiled with TLS support. TLS is not turned on Did anybody experienced with the errors listed below: 08/13 16:28:05 13648 debug <tcp_main.c: 718> tcp_send: no open tcp connection found, opening new one 08/13 16:28:05 13651 err <tcp_main.c: 291> ERROR: tcp_blocking_connect: timeout 10 s elapsed from 10 s 08/13 16:28:05 13651 err <tcp_main.c: 459> ERROR: tcpconn_connect: tcp_blocking_connect failed 08/13 16:28:05 13651 err <tcp_main.c: 721> ERROR: tcp_send: connect failed 08/13 16:28:05 13651 err <../../forward.h: 105> msg_send: ERROR: tcp_send failed 08/13 16:28:05 13651 err <t_fwd.c: 767> ERROR:tm:t_forward_nonack: sending request failed 08/13 16:28:05 13651 debug <t_funcs.c: 255> ERROR:tm:t_relay_to: t_forward_nonack returned error Once I start getting these, proxy will become pretty much un-usable, I have to restart it. Thanks a lot. Hippo -- View this message in context: http://www.nabble.com/openser-1.2.1-errors-in-the-log-tp18973275p18973275.h… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

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[Kamailio-Users] Error: driver error on query: Unknown column 'str_doc_col' in 'field list' ??
by David Loh 14 Aug '08

14 Aug '08

Hi guys, Since 1.4.0 released on Kamailio, I've been testing it on my R&D server. With the introduction of RLS, I'm trying to implement this module along with presence on my box, however, I'm having this error messages whenever rls_handle_subscribe() been called. ------------------------------------------------------------------------ Aug 14 00:06:03 localhost /sbin/kamailio[5139]: INFO: [sip:davidloh@192.168.72.128](192.168.72.1) - [eyeBeam release 1100l stamp 46319] entered Aug 14 00:06:03 localhost /sbin/kamailio[5139]: DBG: Request [SUBSCRIBE] from [sip:davidloh@192.168.72.128] (192.168.72.1) Aug 14 00:06:03 localhost /sbin/kamailio[5139]: _ERROR:db_mysql:db_mysql_submit_query: driver error on query: Unknown column 'str_doc_col' in 'field list'_ Aug 14 00:06:03 localhost /sbin/kamailio[5139]: ERROR:core:db_do_query: error while submitting query Aug 14 00:06:03 localhost /sbin/kamailio[5139]: ERROR:rls:get_resource_list: while querying table xcap for [uri]=sip:davidloh@192.168.72.128 Aug 14 00:06:03 localhost /sbin/kamailio[5139]: ERROR:rls:rls_handle_subscribe: while attepmting to get a resource list Aug 14 00:06:03 localhost /sbin/kamailio[5139]: ERROR:rls:rls_handle_subscribe: occured in rls_handle_subscribe ------------------------------------------------------------------------ Have been google around for the solution but couldn't find any. Appreciate if anyone could enlighten me on this. Meanwhile, I was wondering if any of could enlighten me on how to get Kamailio + Presence + RLS works? No matter how I configure, SIP client (eyeBeam) seems not able to pull out any buddies-list/anything right from XCAP (integrated XCAP from Kamailio). Following is my configuration (unnecessary part have been striped) ------------------------------------------------------------------------ loadmodule "sl.so" loadmodule "tm.so" loadmodule "db_mysql.so" loadmodule "presence.so" loadmodule "presence_xml.so" modparam("presence_xml", "force_active", 1) modparam("presence_xml", "xcap_table", "xcap") modparam("presence_xml", "pidf_manipulation", 0) modparam("presence_xml", "integrated_xcap_server", 1) modparam("presence", "presentity_table", "presentity") modparam("presence", "active_watchers_table", "active_watchers") modparam("presence", "watchers_table", "watchers") modparam("presence", "clean_period", 100) modparam("presence", "server_address", "sip:192.168.72.128:5060") loadmodule "imc.so" loadmodule "mi_xmlrpc.so" modparam("mi_xmlrpc", "log_file", "/var/log/kamailio-xmlrpc.log") modparam("mi_xmlrpc", "port", 8080) loadmodule "pua.so" loadmodule "rls.so" modparam("rls", "integrated_xcap_server", 1) modparam("rls", "xcap_root", "http://192.168.72.128/xcap-root:8080") modparam("rls", "rls_event", "presence") modparam("rls", "to_presence_code", 10) loadmodule "xcap_client.so" modparam("xcap_client", "xcap_table", "xcap") modparam("xcap_client", "query_period", 50) modparam("presence|presence_xml|imc|pua\rls", "db_url", "mysql://myuserid:mypassword@localhost/kamailio") ........... route[0] { .... if (is_method("NOTIFY")) { pua_update_contact(); route(4); } else if (is_method("PUBLISH|SUBSCRIBE")) { route(5); } route[5] { if (!t_newtran()) { sl_reply_error(); exit; }; if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { $var(ret_code) = rls_handle_subscribe(); if($var(ret_code)==10) { handle_subscribe(); } t_release(); } else if(is_method("NOTIFY")) { rls_handle_notify(); xlog("L_INFO", "DBG: RLS handle_notify()"); } exit; } ------------------------------------------------------------------------ Did I missed anything ? Thanks, David Loh

2 1

[Kamailio-Users] Non-numeric prefixes in carrierroute:scan_prefix
by mayamatakeshi 14 Aug '08

14 Aug '08

Hello, I just need a confirmation. We have Kamailio 1.3 running in production and it is working fine. But we received some reports from our OAM team that sometimes they want to set a non-numeric prefix at carrierroute::scan_prefix (like 'abc.') and this seems to not work. I have no complete details but it seems cr_user_rewrite_uri will fail if scan_prefix holds a non-numeric prefix. So I just like to know if scan_prefix is supposed to work with non-numeric prefixes or not. Regards, takeshi

2 1

[Serusers] SER with Radius Authentication
by Samuel Muller 13 Aug '08

13 Aug '08

Hello all, I'm looking for an updated sample of "ser.cfg" who can help me about the use of Radius for authentication. Since release 2.0.0, it seems the function "www_challenge" does not exist ... If anyone has a config part concerning Register + Auth with Radius, it could be very helpfull ! In second point, I just wanna know if a database is always needed for multi-domains environment ? I've some problems with functions like lookup_domain. Many thanks, Samuel MULLER

1 0

[OpenSER-Users] Problem getting redirect reason in uac_redirect
by Ruchir 13 Aug '08

13 Aug '08

I'm using uac_redirect module to handle redirect and accounting from openser. Redirect works fine so as cdr but I'm not getting how to write redirect reason in cdr. I didn't find any way to find and store redirect reason(call forward, busy, no answer) in CDR. Does anyone know how to do it?

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[Kamailio-Users] ERROR:uri_db:mod_init: Invalid table version of the subscriber table (Kamailio ver. 1.4)
by Rafael Risco Gonzales-Vigil 13 Aug '08

13 Aug '08

Hi I am trying to run new kamailio 1.4 but have this errors: *Aug 11 12:56:44 [20808] ERROR:uri_db:mod_init: Invalid table version of the subscriber table Aug 11 12:56:44 [20808] ERROR:core:init_mod: failed to initialize module uri_db Aug 11 12:56:44 [20808] ERROR:core:main: error while initializing modules* I have created new database 'kamailio' using kamdbctl and 'version' table has this values: table_name: subscriber table_version: 6 any idea? rafael PS: same problem with opensips 1.4...

2 4

[Kamailio-Users] listen the new name
by Daniel-Constantin Mierla 13 Aug '08

13 Aug '08

Hello, we did some recordings of the new name, letting you know how some people pronounce the word. The audio files are linked from the first page http://www.kamailio.net/ or can be found directly at: http://www.kamailio.net/pub/kamailio_pronounciation/ Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

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Re: [Kamailio-Users] project foundation
by Daniel-Constantin Mierla 12 Aug '08

12 Aug '08

Hello, On 08/12/08 16:39, Adam Linford wrote: > Good stuff! > > I think that, for the most part, the business and technical facets of > the organisation should remain fairly distinct. The board needs > representation with regards to technical direction (and should have > this input from the organisations official 'members' under my > understanding of the current proposal), but its purpose should be to > keep developers free to code, and users free to use, safe in the > knowledge that resources will be made available (where neccessary) for > the project to achieve its objectives (both long and short term). yes, there are going to be no restrictions in usage or coding, but have an organization with legal framework behind that can take care of trademarks, manage resources and administration of the project. > > We could make non-executive positions on the board available. A good suggestion. > These should be unbiased people.. perhaps invite some big names in the > SIP field, that can provide an independent review of the boards > activities, as well as assisting the board in its role. This is a > tried and tested directorial structure that would also appeal to the > corporate world, which in turn might make it easier to achieve > corporate sponsorship/contributions/membership (as well as adoption of > the software in industry!). > > I don't think an executive committee is needed within the board.. the > board itself should suffice. If it doesn't, we don't have the right > board. I would suggest keeping the main board size small (5-8) to > keep it agile whilst maintaining good representation of members I think would be good to have some clear positions so people should know who to address to when comes to dealing with the foundation. Also, by this, people can allocate the proper time in their schedule for the tasks. We may avoid in this way some members expecting the others to do some work. > > Election rules need discussing.. duration of term for board and also > the achievement of 'member' status.. we dont want potential developers > discouraged from working with us in the traditional open source model > because they would have too much responsibility placed on them to be > 'active members'. Being an active member is just free will. But we want to avoid being a member with voting right just by paying some money. That meber should show interest and contribute to the project in a positive manner. Indeed, this needs to be addressed. > > Frequency of meetings etc are all relatively easy decisions that can > be deferred to the board once i elected. Yes, I believe the same. Cheers, Daniel > > Cheers, > Adam > > On 12 Aug 2008, at 12:16, Daniel-Constantin Mierla wrote: > >> Hello everybody, >> >> I think it is the time to move forward with project management >> structure. During last days there were talks and discussions on the >> lists, the solution to make the project independent and long term >> trustable is to create a foundation. >> >> We made a draft of purpose and organization structure and we want to get >> your comments. >> >> http://www.kamailio.net/dokuwiki/doku.php/admin:foundation-draft >> >> For the statute we are looking at similar cases of foundations around >> open source projects. Considering the developer population and >> contributors, I find as good and reliable place to register the >> foundation to be Germany. >> >> There is still long way to go and a lot to discuss: >> >> * >> >> membership fees >> >> * >> >> how one can become a full member (associate member can be anyone >> that is friend of the project and pays the membership fee) >> >> * >> >> how a member loses the board or full member rights >> >> * >> >> how the board is supervised >> >> o >> >> when and how the reports of the board are presented >> >> o >> >> when and how the general assembly meeting takes place >> >> o >> >> how the reports are approved by full members >> >> * >> >> how to change the statute over time >> >> ... and perhaps more, but if we start and put some effort we can have it >> ready in reasonable time, avoiding trust and reliability issues in long >> term. >> >> Let's have debates here and contribute the results to the wiki. >> >> Cheers, >> Daniel >> >> -- >> Daniel-Constantin Mierla >> http://www.asipto.com >> >> >> _______________________________________________ >> Users mailing list >> Users(a)lists.kamailio.org >> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > -- Daniel-Constantin Mierla http://www.asipto.com

1 0

[Kamailio-Users] branch_route not always called
by Alex Hermann 12 Aug '08

12 Aug '08

Hello again, The uac_replace_from issue led me to another problem. The branch_route is not always called even though I set it using t_on_branch everytime from REQUEST_ROUTE and FAILURE_ROUTE. The failing scenario is as follows: The call is routed to 2 domains (A,B) each with 2 SRV records (1,2). t_on_branch relay domainA branch domainA-1 - timeout branch domainA-2 - 480 failure route append_branch t_on_branch relay domainB branch domainB-1 - timeout branch domainB-2 - 200 OK ==> this branch doesn't call the branch_route Any idea's ? -- Greetings, Alex Hermann

2 3

[Serusers] problem error SER
by Bimantara Jayadiputro 12 Aug '08

12 Aug '08

hello sir i'm bimbim, student from indonesia now i'm trying to finish my final project my final project is about SER i have trouble with that when i start /etc/init.d/ser start there is an error with assigning addres below is the configuration of my ser ------------------------------------------------------------------------------------------------------------------------ cnc9@cnc9-desktop:/var/run/ser$ sudo /etc/init.d/ser start Starting ser: serListening on udp: 10.14.200.95 [10.14.200.95]:5060 udp: 2001:660:4701:1:0:0:0:1 [2001:660:4701:1:0:0:0:1]:5060 udp: sip4.sipv6.net [10.14.200.101]:5060 udp: sip6.sipv6.net [2001:660:4701:1:0:0:0:2]:5060 tcp: 10.14.200.95 [10.14.200.95]:5060 tcp: 2001:660:4701:1:0:0:0:1 [2001:660:4701:1:0:0:0:1]:5060 tcp: sip4.sipv6.net [10.14.200.101]:5060 tcp: sip6.sipv6.net [2001:660:4701:1:0:0:0:2]:5060 Aliases: tcp: localhost:5060 tcp: cnc9-desktop.local:5060 udp: localhost:5060 udp: cnc9-desktop.local:5060 stateless - initializing 0(0) . cnc9@cnc9-desktop:/var/run/ser$ Maxfwd module- initializing textops - initializing 0(0) INFO: udp_init: SO_RCVBUF is initially 109568 0(0) INFO: udp_init: SO_RCVBUF is finally 219136 0(0) INFO: udp_init: SO_RCVBUF is initially 109568 0(0) INFO: udp_init: SO_RCVBUF is finally 219136 0(0) INFO: udp_init: SO_RCVBUF is initially 109568 0(0) INFO: udp_init: SO_RCVBUF is finally 219136 * 0(0) ERROR: udp_init: bind(6, 0x80f8d7c, 16) on 10.14.200.101: Cannot assign requested address* ------------------------------------------------------------------------------------------------------------------------ i also attached my ser.cfg i hope you can help me resolve this problem thank you

3 2

[Kamailio-Users] uac_replace_from and lumps again
by Alex Hermann 12 Aug '08

12 Aug '08

Hello, i'm still fighting with the very annoying lump model that OpenSER uses. I need to send out INVITE's with different From headers in a failover situation. More specifically I need to send out each step -original request and failure_routes- with a different From header. If any request forks (parallel or serial due to DNS SRV or registered contacts), it needs to keep current (same) the replacement From on all branches. I tried the following: try 1: REQUEST_ROUTE: uac_replace_from(a) t_relay(); FAILURE_ROUTE: append_branch() uac_replace_from(b) t_relay() the second uac_replace_from always produces garbage in the From try 2: REQUEST_ROUTE: t_newtran() uac_replace_from(a) t_relay(); FAILURE_ROUTE: append_branch() uac_replace_from(b) t_relay() Now the second uac_replace_from is correct, but when the first t_relay() does dns based failover, the original from is used for the second branch instead of the replacement. If the response code is locally generated 408, the second uac_replace_from produces garabage too. Is someone aware of a working solution for this (very common) situation? -- Greetings, Alex Hermann

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[Kamailio-Users] Need Help Configuring OpenSer with OpenXCAP
by Noble Antony T 12 Aug '08

12 Aug '08

Hi, I could configure Openser as RLS, and installed OpenXCAP. I wanted to use OpenXCAP with EyeBeam Client. (version 1.5.10.2 build 33793) OpenXCAP is giving the following error. Invalid Document Selector xcap-root(a)192.168.176.150/pres-rules/users/user1/presrules (the Document Selector context must be 'users' or 'global': 'pres-rules') Invalid Document Selector xcap-root(a)192.168.176.150/resource-lists/users/user1/resource-list.xml (the Document Selector context must be 'users' or 'global': 'resource-lists') Invalid Document Selector xcap-root(a)192.168.176.150/pres-rules/users/user1/presrules (the Document Selector context must be 'users' or 'global': 'pres-rules') Invalid Document Selector xcap-root(a)192.168.176.150/resource-lists/users/user1/resource-list.xml (the Document Selector context must be 'users' or 'global': 'resource-lists') UE is getting 404 Error. It seems the OpenXCAP is expecting 'users' or 'global' instead of 'resource-lists'. Can somebody please point out, if I am missing something..? I am attaching my Config Files along with this. Somebody please help me... -- Thanks & Regards, Noble Antony Thattil.

1 1

[Kamailio-Users] fr_inv_timer increased to 120 secs
by Kaeman Chris 12 Aug '08

12 Aug '08

Hello, What are the problems in production that can be encountered if the fr_inv_timer is increased from 30-45 secs to 120 secs for a sip-sip call? Anyone have found any issues? Thanks! KC --------------------------------- Unlimited freedom, unlimited storage. Get it now

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Re: [Kamailio-Users] ERROR:presence:mod_init: 1Wrong version v-1 for table <presentity>, need v2
by Daniel-Constantin Mierla 12 Aug '08

12 Aug '08

Hello, On 08/12/08 10:30, mahesh.peddi(a)info-spectrum.com wrote: > Hi Daniel, > > Thanks for your prompt reply.The output of "show create > table version" is as follows > > +---------+--------------------------------------------------------------------------------------------------------------------------------------------------------------+ > > | Table | Create Table | > +---------+--------------------------------------------------------------------------------------------------------------------------------------------------------------+ > > | version | CREATE TABLE `version` ( > `table_name` varchar(32) NOT NULL, > `table_version` int(10) unsigned NOT NULL default '0' > ) ENGINE=MyISAM DEFAULT CHARSET=latin1 | > +---------+--------------------------------------------------------------------------------------------------------------------------------------------------------------+ > > 1 row in set (0.03 sec) > > Kindly guide me on this. the table structure is ok. Could you make sure the mysql module is compiled again if you updated libraries or so. I was pointed that there was a change in naming the INT type in mysql libraries -- maybe others on the list can jump quicker here with a yes or no. Cheers, Daniel > > Thank you, > > > Regards, > Mahesh Peddi > Infospectrum India Pvt. Ltd. > Cell: +91 9765775176 > IP-Phone Ext. - 764 > > ----- Original Message ----- From: "Daniel-Constantin Mierla" > <miconda(a)gmail.com> > To: <mahesh.peddi(a)info-spectrum.com> > Cc: <users(a)lists.kamailio.org> > Sent: Monday, August 11, 2008 9:41 PM > Subject: Re: [Kamailio-Users] ERROR:presence:mod_init: 1Wrong version > v-1 for table <presentity>, need v2 > > >> Hello, >> >> please post the output of: >> >> show create table version >> >> in the mysql client. The errors shows some issue with the column type. >> >> Cheers, >> Daniel >> >> On 08/11/08 17:12, mahesh.peddi(a)info-spectrum.com wrote: >>> Hi All, >>> >>> I am working on Openser 1.3.2 version with no_tls support. >>> When i include >>> "modparam("presence", >>> "db_url","mysql://openser:openserrw@localhost/openser")" >>> >>> for the presence and run openserctl moni i am getting the error as >>> >>> ERROR: Error opening OpenSER's FIFO /tmp/openser_fifo >>> ERROR: Make sure you have the line 'modparam("mi_fifo", "fifo_name", >>> "/tmp/openser_fifo")' in your config >>> ERROR: and also have loaded the mi_fifo module. >>> >>> And the errors in /var/log/messages >>> >>> Aug 10 09:37:41 rhel4 openser[5530]: ERROR:core:table_version: >>> invalid type (2) or nul (0) version columns for presentity >>> Aug 10 09:37:41 rhel4 openser[5530]: ERROR:presence:mod_init: 1Wrong >>> version v-1 for table <presentity>, need v2 >>> Aug 10 09:37:41 rhel4 openser[5530]: ERROR:core:init_mod: failed to >>> initialize module presence >>> Aug 10 09:37:41 rhel4 openser[5530]: ERROR:core:main: error while >>> initializing modules >>> Aug 10 09:37:41 rhel4 openser[5530]: NOTICE:presence:destroy: >>> destroy module ... >>> Aug 10 10:01:01 rhel4 crond(pam_unix)[5552]: session opened for user >>> root by (uid=0) >>> Aug 10 10:01:01 rhel4 crond(pam_unix)[5552]: session closed for user >>> root >>> >>> And when i check in the data base its showing the version as >>> >>> mysql> select * from version where table_name="presentity" ; >>> +------------+---------------+ >>> | table_name | table_version | >>> +------------+---------------+ >>> | presentity | 2 | >>> +------------+---------------+ >>> 1 row in set (0.00 sec) >>> >>> Please help me how to resolve this. >>> And as mention in the forums i modify the res.c as below shows even >>> i am facing the same error. >>> >>> >>> --------------------------------------------------------------------------------Index: >>> modules/mysql/res.c===================================================================--- >>> >>> modules/mysql/res.c (revision 4448)+++ modules/mysql/res.c (working >>> copy)@@ -103,7 +103,10 @@ break; default:- RES_TYPES(_r)[i] = >>> DB_STRING;+ LM_WARN("unhandled data type column (%.*s) type id (%d), >>> "+ "use DB_STRING as default\n", RES_NAMES(_r)[col]->len,+ >>> RES_NAMES(_r)[col]->s, fields[col].type);+ RES_TYPES(_r)[col] = >>> DB_STRING; break; } >>> }--------------------------------------------------------------------------------I >>> am enclosing my "openser.cfg" and "openserctlrc"Regards,Mahesh >>> PeddiInfospectrum India Pvt. Ltd.Cell: +91 9765775176IP-Phone Ext. >>> - 764 >>> ------------------------------------------------------------------------ >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users(a)lists.kamailio.org >>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >>> >> >> -- >> Daniel-Constantin Mierla >> http://www.asipto.com >> >> > -- Daniel-Constantin Mierla http://www.asipto.com

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Re: [Kamailio-Users] poll: project name
by Helmut Kuper 12 Aug '08

12 Aug '08

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi Klaus, Of course I agree. It's a matter of time to know what Kamailio stand for. I think Apache is a nicer name than Kamailio, but anyway: I agree the name is mainly a question of personal taste and currently I don't like that taste - nevertheless, I think the name should be changed to a more handy one, just to round up the whole product. And sure: The name won't stop telcos from using the Software rather that project structure, roadmap and quality. regards and peace Helmut Klaus Darilion schrieb: | Helmut Kuper wrote: |> -----BEGIN PGP SIGNED MESSAGE----- |> Hash: SHA1 |> |> Hm, |> |> well, maybe I have to make it a bit more clear what I meant. Kamailio is |> pronounceable in german, but it is not easy in terms of smoothyness - |> like openser was or freeswitch is ... |> |> KAMAILIO hmmm no, not with the best will in the world: I have problems |> with it to say it without explaining what that name points to. I think |> it is not a good name to inspire confidence. Maybe it needs a slogan to |> explain a bit more :D | | IMO this is all a matter of time. Would you think that Apache is a web | server and Eclipse is an IDE and Asterisk is an IP PBX? Me not. But now, | if you tell one of these names to an IT guy they know what you are | talking about. Thus, in my opinion, we just need some trust in the | product - I do not believe the argument that telcos will use open source | only if the name indicates open/free/SIP.... Nobody uses a product just | because of its name. | | But I have to accept that others don't like name ...:-( | | regards | klaus | | -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.8 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iEYEARECAAYFAkihQDkACgkQ4tZeNddg3dwrJgCdENGNK/PAid+ftRtvgUFyP9hY ZXAAoJ5ErmBLbKuxTRvp6WmWzudI1BS9 =cyPy -----END PGP SIGNATURE-----

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[Kamailio-Users] Identifying branch in ONREPLY-ROUTE?
by Alex Balashov 12 Aug '08

12 Aug '08

What is the simplest way to identify the branch (by branch index) to which a reply belongs in an ONREPLY-ROUTE? I have parallel forking with my registrar and do passive call statekeeping. The problem is that it must be branch-aware; I need to know if, for example, I got a successful provisional reply on one branch of the call and a failure on the other, and deal with those separately. Right now my statekeeping information is keyed by Call-ID, but since that persists across two branches, obviously some sort of branch-identifying information has to be present. I've tried setting a persistent variable to the value of $T_branch_idx inside a BRANCH-ROUTE, but in ONREPLY-ROUTE that just shows up with the most recent branch index, regardless of which branch the reply actually came in on. I'm probably missing something quite obvious here, so I ask - how do I incorporate some form of branch awareness into ONREPLY-ROUTEs? I really do not care what form that is - be it branch index, or some other built-in value, as long as it is consistent. -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599

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Re: [Kamailio-Users] simple question: log 180, 200 responses
by Alex Balashov 12 Aug '08

12 Aug '08

Ali Soltani wrote: > Hi Alex, > > Thank you for the response. > > I tried to use onreply_route block in the main route as well as in > route1 as follow but still can not log response messages (180, 200, ..) > > route > { > t_on_reply("1"); > ... > } > > route[1] > { > if (is_method("INVITE")) > { > t_on_reply("1"); > } > ... > } > > onreply_route[1] > { > xlog("(onreplyroute1) usr=$ru method=$rm request_port=$rp > received_port=$Rp original port=$op destination_port=$dp src_port=$sp"); > } You can only have one reply route per request. If you specify multiple ones, only the last one will take effect. The problem in this case is that route[1] will never be called. Here's what you want: route { if(is_method("INVITE")) t_on_reply("1"); } onreply_route[1] { ... } -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599

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[Kamailio-Users] free_fixup_function
by Julien mangeard 12 Aug '08

12 Aug '08

In the 1.4 release notes, I have read that: => new free fixup functions - making safer to use some functions exported by modules in embedded languages such as Perl I don't find any reference to "free_fixup_function" in the Perl module source code. Are you sure that this feature works for Perl module? Someone work on it? Best Regards, -- Julien Mangeard

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[Serusers] Fw: 20080812 - Need help with configuring SER as a call forwarder server
by Morteza Alizadeh 12 Aug '08

12 Aug '08

Dear Sir/Madam, My previous email has been held by your system due to its large volume. I removed the log file from it and resend it again. If you think, that logs of the SER would help you, please provide me the way to send it as well. Your help is much appreciated. --- Sincerely yours, Morteza Alizadeh Design Engineer Cell Phone: +60 (12)-319-7439 Email: malizadeh(a)emerj.com Morteza.Alizadeh(a)gmail.com Skype: mortezaalizadeh Block D-17-6, Gurney Height Condominium, Jalan Bukit Keramat, 54000 Kuala Lumpur, Malaysia ----- Forwarded Message ---- From: Morteza Alizadeh <malizadeh(a)e-merj.com> To: serusers(a)lists.iptel.org Cc: vazirif(a)e-merj.com Sent: Tuesday, August 12, 2008 10:09:35 AM Subject: 20080812 - Need help with configuring SER as a call forwarder server Dear Sir/Madam, I am trying to use and configure your SER in a manner to let us forward our landline calls to the public sip.broadvoice.com gateway. Unfortunately, I am completely mixed up with the way it should be setup. There are some questions that I would be really appreciated if you can answer them or forward it to a right person. Please consider that I have already fully configured your SER to handle internal IP call between SIP clients and also I tested the SER with the SIP clients which are behind NAT demonstrate its NAT module works fine. Here is my current setup parameter to access landline call: * SER IP address: 207.34.42.53 * SER domain: ubuntu-serv.emerj.net * SIP gateway: sip.broadvoice.com * Internal user ID: 10002 * BroadVoice contact: 7184122016 * Number wants to access through SER: 9178462343 What I have in my MySQL server is as follow: * Domain ubuntu-serv.emerj.net in table 'domain'. * User 10002 registered with domain ubuntu-serv.emerj.net in table 'subscriber'. The password for this account is the normal password I set for it in database. * User 7184122016 registered with domain sip.broadvoice.com in table 'subscriber'. The password for this account is the password has been given by BroadVoice to us. * Defined a record in table 'usr_preferences' contains the following information: +------+------------+--------+-----------+-----------------------------------+------+--------------------------------------------+ | uuid | username | domain | attribute | value | type | modified | +------+------------+--------+-----------+-----------------------------------+------+--------------------------------------------+ | | 9178462343 | | callfwd | sip:7184122016@sip.broadvoice.com | 0 | 2008-08-11 10:00:29 | +------+------------+--------+-----------+-----------------------------------+------+--------------------------------------------+ This is the information I am using for my call forwarding purpose. Whenever the user tries to dial 9178462343, this record blindly forward it to the BroadVoice.com. Here are my questions: 1. I monitored and captured the packet between SIP, SER and BroadVoice. For the normal call directly from SIP to BroadVoice and also between SIP client and SER, a kind of SIP REGISTER message is sent at initial time and it seems that all the rest messages is based on the session value that is initially exchanged. When I try to use my SER as a call forwarding server, and when my SIP client tries to dial, SER doesn't send REGISTER message for account to 7184122016 to the BroadVoice. I want to know whether it is possible for SER to do this REGISTER before sending INVITE message for 9178462343 or not. 1. Is the call forwarding is the correct solution for this issue or not? Maybe there are some other things that I am not aware of them due to I am really new to SER environment. 1. I also saw that SER doesn't overwrite the values of Contact, To, and From in the INVITE message properly and everything is sent to the BroadVoice with the values of the ubuntu-serv.emerj.net as a domain and also initial SIP contact that all of them are unknown for BroadVoice gateway. How should I tell SER to overwrite this values and keeps the track of the changes? I also attached my both ser.cfg and the logs of the SER with my email. I would be really appreciated if you assist me resolving this issue and helping me with any document or sample ser.cfg file showing me how to realize the idea. I am looking forward to you response. --- Sincerely yours, Morteza Alizadeh Design Engineer Cell Phone: +60 (12)-319-7439 Email: malizadeh(a)emerj.com Morteza.Alizadeh(a)gmail.com Skype: mortezaalizadeh Block D-17-6, Gurney Height Condominium, Jalan Bukit Keramat, 54000 Kuala Lumpur, Malaysia

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[Kamailio-Users] A function to add uri/header parameters
by Iñaki Baz Castillo 11 Aug '08

11 Aug '08

Hi, for now, the "normal" way to detect if an in-dialog request/response beyonds to a natted dialog is by adding a uri parameter ("nat=yes") in the "Contact" header of the first request and response and macthing that parameter in the RURI of subsequent in-dialog requests. Typically this is done in this way: search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes'); Well, this mechanims (that is part of official OpenSer courses) is very RCF3261 non compliant: 1) In SIP a header field name is case insensitive but the above function wouldn't match a "CONTACT" of "contact" header names (even if they are 100% valid). 2) "Contact" header can appear abbreviated as "m" (or "M") but the above doesn't match it. 3) 1.1.1. Known Limitations search ignores folded lines. For example, search("(From|f):.*@foo.bar") doesn't match the following From header field: From: medabeda <sip:medameda@foo.bar>;tag=1234 A solution for points 1 and 2 could be using "subst" with insensitive flag enabled, but it could be very anti-friendly. So I wonder: wouldn't be useful a good funtion to add/read/remove header and uri parameters (in any header or uri) being RFC 3261 compliant (this is: case insensitive and allowing header name abbreviations)? -- Iñaki Baz Castillo

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[Kamailio-Users] ISN routing
by Krunal Patel 11 Aug '08

11 Aug '08

Hi, I would like to implement isn routing in openser. I am new to isn routing. Please help me if anybody of you have idea about isn routing. Thanking in advance - Krunal Patel

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[Kamailio-Users] ERROR:presence:mod_init: 1Wrong version v-1 for table <presentity>, need v2
by mahesh.peddi＠info-spectrum.com 11 Aug '08

11 Aug '08

Hi All, I am working on Openser 1.3.2 version with no_tls support. When i include "modparam("presence", "db_url","mysql://openser:openserrw@localhost/openser")" for the presence and run openserctl moni i am getting the error as ERROR: Error opening OpenSER's FIFO /tmp/openser_fifo ERROR: Make sure you have the line 'modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo")' in your config ERROR: and also have loaded the mi_fifo module. And the errors in /var/log/messages Aug 10 09:37:41 rhel4 openser[5530]: ERROR:core:table_version: invalid type (2) or nul (0) version columns for presentity Aug 10 09:37:41 rhel4 openser[5530]: ERROR:presence:mod_init: 1Wrong version v-1 for table <presentity>, need v2 Aug 10 09:37:41 rhel4 openser[5530]: ERROR:core:init_mod: failed to initialize module presence Aug 10 09:37:41 rhel4 openser[5530]: ERROR:core:main: error while initializing modules Aug 10 09:37:41 rhel4 openser[5530]: NOTICE:presence:destroy: destroy module ... Aug 10 10:01:01 rhel4 crond(pam_unix)[5552]: session opened for user root by (uid=0) Aug 10 10:01:01 rhel4 crond(pam_unix)[5552]: session closed for user root And when i check in the data base its showing the version as mysql> select * from version where table_name="presentity" ; +------------+---------------+ | table_name | table_version | +------------+---------------+ | presentity | 2 | +------------+---------------+ 1 row in set (0.00 sec) Please help me how to resolve this. And as mention in the forums i modify the res.c as below shows even i am facing the same error. --------------------------------------------------------------------------------Index: modules/mysql/res.c===================================================================--- modules/mysql/res.c (revision 4448)+++ modules/mysql/res.c (working copy)@@ -103,7 +103,10 @@ break; default:- RES_TYPES(_r)[i] = DB_STRING;+ LM_WARN("unhandled data type column (%.*s) type id (%d), "+ "use DB_STRING as default\n", RES_NAMES(_r)[col]->len,+ RES_NAMES(_r)[col]->s, fields[col].type);+ RES_TYPES(_r)[col] = DB_STRING; break; } }--------------------------------------------------------------------------------I am enclosing my "openser.cfg" and "openserctlrc"Regards,Mahesh PeddiInfospectrum India Pvt. Ltd.Cell: +91 9765775176IP-Phone Ext. - 764

2 1

[Kamailio-Users] naming of open source projects
by Klaus Darilion 11 Aug '08

11 Aug '08

Hi! I really do not understand why so many people think that the name should include the word "open" or should somehow include parts of the protocol (SIP) or its usage (server, proxy ...). If I take a look at very successful open source projects they do not follow this, e.g. Apache, Eclipse, Pidgin, Wireshark, Asterisk, Tomcat, Boost ... regards Klaus

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Re: [Kamailio-Users] poll: project name
by luzango mfupe 11 Aug '08

11 Aug '08

"Free to Talk" Not bad at all i bet most of us here would prefer an English name!!! Why Kamailio is bad??? relax guys u will be able to pronounce it well soon... My choice iiis numberrr 2. On Mon, Aug 11, 2008 at 3:31 PM, <users-request(a)lists.kamailio.org> wrote: > Send Users mailing list submissions to > users(a)lists.kamailio.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > or, via email, send a message with subject or body 'help' to > users-request(a)lists.kamailio.org > > You can reach the person managing the list at > users-owner(a)lists.kamailio.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Users digest..." > > > Today's Topics: > > 1. Re: [Kamailio-Devel] poll: project name (Klaus Darilion) > 2. Re: poll: project name (Yazid Hadj Said) > 3. Re: poll: project name (Ardjan Zwartjes) > 4. Re: [Kamailio-Devel] poll: project name (David Villasmil) > 5. Re: [Kamailio-Devel] poll: project name ( I?aki Baz Castillo ) > 6. Re: poll: project name (Carsten Bock) > 7. Re: poll: project name (Alex Balashov) > 8. Re: poll: project name (Johansson Olle E) > 9. Re: [Kamailio-Devel] poll: project name (David Villasmil) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Mon, 11 Aug 2008 13:39:29 +0200 > From: Klaus Darilion <klaus.mailinglists(a)pernau.at> > Subject: Re: [Kamailio-Users] [Kamailio-Devel] poll: project name > To: miconda(a)gmail.com > Cc: users <users(a)lists.kamailio.net>, devel <devel(a)lists.kamailio.net> > Message-ID: <48A024F1.5030105(a)pernau.at> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > speaking as user: > 1) Yes, I love it. Please keep it. > > Daniel-Constantin Mierla schrieb: > > Hello everybody, > > > > some people expressed inconveniences regarding the new name, so we would > > like to get your opinion about, therefore this is a poll that will get > > us to a decision. Here is the question: > > > > Do you like the name "Kamailio"? > > 1) Yes, I love it. Please keep it. > > 2) It is OK. > > 3) I do not mind the name at all > > 4) I do not like it > > 5) I hate the name and we should change it > > > > There is no need of explanations, as it is up to personal taste and > > decision. Just vote by option. > > > > Cheers, > > Daniel > > > > PS. for those interested... > > > > As short history about renaming story. On the 28th of March the board > > was informed about issues with SER part of project name. You can easily > > find who has the trademarks. On the 7th of April the board was informed > > about an agreement for the deadline as of September 30, 2008 to use old > > name. From there on, all old releases, documents, etc. can be referred > > by the old name, but nothing new. Other few discussions about what > > should/could be kept/changed followed and on 12th of May started the > > discussion about the new name with two main direction, something > > similar, to remind about old project or just something completely new, > > to be used as brand name. Couple of proposals and discussions regarding > > possible names followed up to 16th of July when was the deadline and the > > start of vote. Majority was reached on 17th but it was required to have > > opinions from all members of the board, last came from Bogdan, on the > > 28th of July. > > > > All discussions about migration were on the board mailing list. No > > change were done to openser.org configuration, just the new domain with > > kamailio in it was pointed to same server. Every step done was reported > > to board, following a schedule discussed there. Another discussion was > > about making the renaming and 1.4.0 release in same day. Being > > practically impossible to do all the work in one day, it was decided to > > do first the announcement about renaming. That happened on 28th when all > > agreed with the new name. > > > > As you can see, there was no 5 months of renaming duration. There was no > > change in openser.org DNS configuration undertaken with/without the > > knowledge of all board members. The new name domains were reserved when > > majority was achieved, in any case; dns entries for new name were > > pointed to old server shortly after booking the domains. > > > > > > ------------------------------ > > Message: 2 > Date: Mon, 11 Aug 2008 13:52:02 +0200 > From: "Yazid Hadj Said" <yazid.hadjsaid(a)gmail.com> > Subject: Re: [Kamailio-Users] poll: project name > To: users(a)lists.kamailio.net > Message-ID: > <8dfa8c1a0808110452w274cb206te3160cbe4dd8fd2(a)mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > Hi all, > > > >Do you like the name "Kamailio"? > >1) Yes, I love it. Please keep it. > >2) It is OK. > >3) I do not mind the name at all > >4) I do not like it > >5) I hate the name and we should change it > > I do not like it. > I think that at least the word "Open" must be included in the new name > > > Regards > Yazid >

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Re: [Kamailio-Users] simple question: log 180, 200 responses
by Yazid Hadj Said 11 Aug '08

11 Aug '08

Hi Ali, I think that you must call : if(status=="180") { xlog("this is a 180 Ringing reply"); }; if(status=="200") { xlog("this is a 200 OK reply"); }; in the onreply_route, note in tne route function

1 0

[Kamailio-Users] simple question: log 180, 200 responses
by Ali Soltani 11 Aug '08

11 Aug '08

Hi, I am using openSER as a simple sip proxy for calls between two sip end-points. By snooping the traffic I can verify that the openSER proxies ALL sip messages/responses but non of the following log statements do not log the response messages (180, 200). Am I missing anything. route { xlog("(route) usr=$ru method=$rm request_port=$rp received_port=$Rp original port=$op destination_port=$dp src_port=$sp"); xlog("(route) reply_status=$rs"); if(status=="180") { xlog("this is a 180 Ringing reply"); }; if(status=="200") { xlog("this is a 200 OK reply"); }; ..... } Thanks,

2 1

Re: [Kamailio-Users] poll: project name
by Yazid Hadj Said 11 Aug '08

11 Aug '08

Hi all, >Do you like the name "Kamailio"? >1) Yes, I love it. Please keep it. >2) It is OK. >3) I do not mind the name at all >4) I do not like it >5) I hate the name and we should change it I do not like it. I think that at least the word "Open" must be included in the new name Regards Yazid

1 0

[Serusers] 404 Not Found
by dwianto dana 11 Aug '08

11 Aug '08

Hi, I tried to use SER openimscore and connect it with sailfin. Tried to register using X-Lite. What I get in the Core is SIP: Status:404 Not Found - destination user not found on this S-CSCF I am using a default user (Alice and Bob),, so it is kinda strange to get this information, user not found. Thanks for the answers given

1 0

[Kamailio-Users] Req for Coments about my config
by Ali Jawad 11 Aug '08

11 Aug '08

Dear All I am working with some developers to create a sip client that supports TLS for me. During the beta stage they created the phone and it sent "sips" packets with transport=tls in the packet. The thing is that I used the sethostport and t_relay to send data to my PSTN gateway that works on UDP but in both cases I was not able to make PSTN calls. The developers created a custom openser.cfg that rewrites sips packets to sip and removes the transport=tls from the packets. How efficient is this ? Is this the typical approach to make a TLS call through Openser where the PSTN gateway does not support TLS ? Please have a look at the config file if you can. http://pastebin.com/m20f7d25a Thanks

2 1

Re: [Kamailio-Users] openser/kamailio meeting in Bucharest, August 11, 2008
by Cesar Troncoso 11 Aug '08

11 Aug '08

Hello! I am also up for a meeting in Bucharest on Monday next week. It will be a good chance to get different points of view about the kamailio/opensips discussion and the way developers and board members think to move forward in the future. Regards, Cesar ------------------------------ Message: 6 Date: Tue, 05 Aug 2008 18:57:50 +0300 From: Daniel-Constantin Mierla <miconda(a)gmail.com> Subject: [Kamailio-Users] openser/kamailio meeting in Bucharest, August 11, 2008 To: users <users(a)lists.kamailio.org>, devel <devel(a)lists.kamailio.org>, business(a)lists.kamailio.org Message-ID: <4898787E.7020602(a)gmail.com> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Hello Everybody, as many expressed their intention for an openser/kamailio event in Bucharest, next week, I am going to organize a dinner in a place with nice Romanian beer, wine and food. Seeing the suggestions for the date, I am proposing August 11, 2008 (Monday) -- let me know if you have opinions. Exact hour and place will be sent as soon as I get an idea about number of participants. So far, expressed the intention to participate: - Henning Westerholt, developer and board member - Elena-Ramona Modroiu, developer and board member - Daniel-Constantin Mierla, developer and board member - Michiel ten Hagen - Sebastian Schuman Please write me if you want to attend, I need to know the number of attendants to make the reservation. The event is in the same fashion as the ones happening in Barcelona, Vienna, Berlin, ..., in the past -- everybody pays for its expenses. If you want to have some preamble (might not be same place, but...), you can check some photos from last similar event in Bucharest after the ROSDEV2008 conference: http://www.asipto.com/photos/ROSDEV2008-1-ALBUM/index1.html Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

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[Kamailio-Users] fr_inv_timer issue
by Kaeman Chris 11 Aug '08

11 Aug '08

Hello, We have an fr_inv_timer set to 45 secs. When i make calls to an 800 number, the call hits to the 1st provider on LCR and IVR recording starts asking to hit an option 1,2,3,etc & if the caller dont hit any option and leave the call up to 45 secs. Than fr_inv_timer kills the transaction by sending a CANCEL message and thereby openser forces the call to send it over to the fail over route. Is this correct should it try the call on the failover route because the 1st provider was able to complete the call than why the call went over to the failover route? here as soon as the call switches to the 2nd provider the IVR message starts again from the option 1 leaving the option 3 in the middle of the recording. is this correct? let me know if the query is not clear. Thanks in advance --------------------------------- Unlimited freedom, unlimited storage. Get it now

3 2

[Kamailio-Users] Radius + openser
by Amit Jain 11 Aug '08

11 Aug '08

Hello everyone, I was looking for documentation for Radius AAA + Openser. I tried to google but the links provided are not working. i think since the site has been renamed. Can any provide me document if saved or any other related link. Link to which i was referring . http://openser.org/docs/openser-radius-1.0.x.html Regards, Amit Jain

2 2

[Kamailio-Users] Radius + openser
by Amit Jain 11 Aug '08

11 Aug '08

Hello everyone, I was looking for documentation for Radius AAA + Openser. I tried to google but the links provided are not working. i think since the site has been renamed. Can any provide me document if saved or any other related link. Link to which i was referring . http://openser.org/docs/openser-radius-1.0.x.html Immediate help required :( Regards, Amit Jain

2 1

[Kamailio-Users] Should really 't_on_branch' remain in serial forking?
by Iñaki Baz Castillo 11 Aug '08

11 Aug '08

Hi, imagine this simple case: -------------------------- route { t_on_branch("1"); t_on_failure("2"); xlog("We are in 'route'"); t_relay("1.1.1.1"); } branch_route[1] { xlog("We are in 'branch_route[1]'"); # do something.... } failure_route[2] { xlog("We are in 'failure_route[2]'"); append_branch(); t_relay("2.2.2.2"); } --------------------------- In case an error occurs when forwarding the request to 1.1.1.1 this would be the screen log output: We are in 'route' We are in 'branch_route[1]' We are in 'failure_route[2]' We are in 'branch_route[1]' This is: branch_route[1] will also be runned **again** after the failure route since it was loaded in the first forward attemp. I really don't know if this is intuitive or not. The only way to "dissable" branch_route[1] in the failure route is by adding: ------------------- failure_route[2] { t_on_branch("2"); # <--- Dissable t_on_branch("1") xlog("We are in 'failure_route[2]'"); append_branch(); t_relay("2.2.2.2"); } branch_route[2] { xlog("We are in 'branch_route[2]'"); # Nothing to do here. # This route is neccesary to dissable t_on_branch("1") } ------------------ This would show: We are in 'route' We are in 'branch_route[1]' We are in 'failure_route[2]' We are in 'branch_route[2]' What about if "t_on_branch" wouldn't remain loaded after a failure route or serial forking? wouldn't be more intuitive to re-enable it explicitely when required? -- Iñaki Baz Castillo

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[Kamailio-Users] Res: serMyAdmin login problem invalid username and/or password
by Flavio Goncalves 10 Aug '08

10 Aug '08

Hi Muhammad, I will check the problem this week and return to you soon. Regards, Flavio ----- Mensagem original ---- De: muhammad akl <muhammad.akl(a)gmail.com> Para: Flavio Goncalves <flavio(a)asteriskguide.com> Cc: users(a)lists.kamailio.org Enviadas: Domingo, 10 de Agosto de 2008 12:44:47 Assunto: Re: [Kamailio-Users] serMyAdmin login problem invalid username and/or password thx Flavio for ur response but i still have another question the main problem with the user "admin@localhost " and password "openserrw" that were created by the openser_mysql.sh is still happening got the same message "invalid username and/or password " but when i removed that account from the subscriber table and used the openser.sql file from sermyadmin i have successfully logged in but after modifying the openser.sql file to match my environment so after that i tried to register with a new account from the home page using register button besides to login button and filled the account's information there everything worked fine but the problem happend here now is when i logged with the "admin@setup" and password "secret " account to approve the new account for users i got this message pastebin.com/m7154020b p.s for alot of testing i have used more than one version of serMyAdmin beginning with ver 0.3 to 0.7 !! sorry for my poor english as am not native BR muhammad

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[Kamailio-Users] Res: serMyAdmin login problem invalid username and/or password
by Flavio Goncalves 10 Aug '08

10 Aug '08

Hi Muhammad, Check the updated procedure at www.sermyadmin.org. The default login is asmin@setup, password secret as stated in the procedure. Flavio E. Goncalves ----- Mensagem original ---- De: muhammad akl <muhammad.akl(a)gmail.com> Para: users(a)lists.kamailio.org Enviadas: Domingo, 10 de Agosto de 2008 6:28:53 Assunto: [Kamailio-Users] serMyAdmin login problem invalid username and/or password hey all i'm trying to configure sermyadmin web interface to administrate openser-1.2.2 through it firstly i installed apache-tomcat and configure it successfully then i downloaded the sermyadmin.war and copied it to /usr/local/tomcat/webapps then downloaded mysql-jdbc and copied it to /usr/local/tomcat/lib also configure the context.xml and modified the Context element and added the user have full privileges to access openser databases then when tried to login with user "admin@locahost" and pass " openserrw" default values created by the openser_mysql.sh i got invalid username or password ? any suggestions ? serMyAdmin version used .7-BETA i also tried the howto found in sermyadmin.org but with no success by trying to use the prepared openser.sql file if anyone could i'd appreciate that

2 1

[Kamailio-Users] How to revert changes in SDP during serial forking?
by Iñaki Baz Castillo 10 Aug '08

10 Aug '08

Hi, I have a scenario in which the call is first forwarded to user location using MediaProxy (so I modify the SDP in OpenSer) and if it gets a error response a failure route forwards the request to a media server with public IP for which I don't need and don't want to use MediaProxy. But since I change the SDP in the first forwarding (to the user) then I can't use the original SDP in the sequential forwarding to the media server. Is possible to do it? Thanks a lot. -- Iñaki Baz Castillo

2 2

[Kamailio-Users] serMyAdmin login problem invalid username and/or password
by muhammad akl 10 Aug '08

10 Aug '08

hey all i'm trying to configure sermyadmin web interface to administrate openser-1.2.2 through it firstly i installed apache-tomcat and configure it successfully then i downloaded the sermyadmin.war and copied it to /usr/local/tomcat/webapps then downloaded mysql-jdbc and copied it to /usr/local/tomcat/lib also configure the context.xml and modified the Context element and added the user have full privileges to access openser databases then when tried to login with user "admin@locahost" and pass " openserrw" default values created by the openser_mysql.sh i got invalid username or password ? any suggestions ? serMyAdmin version used .7-BETA i also tried the howto found in sermyadmin.org but with no success by trying to use the prepared openser.sql file if anyone could i'd appreciate that

1 0

[Kamailio-Users] OpenSIPS project
by Bogdan-Andrei Iancu 10 Aug '08

10 Aug '08

Hello everybody, A new project - OpenSIPs (Open SIP Server) - became active today, with its first release (OpenSIPS 1.4.0). What is OpenSIPS about? OpenSER History: ---------------- Voice System started the OpenSER project with a vision of running an open project that will deliver a professional software. Along the years Voice System was the main contributor and sustainer of the OpenSER project, organizing conferences, events and tutorials, all aiming to consolidate OpenSER as a top solution for industry VoIP. OpenSER Issues: --------------- In the past time, several critical problems related to OpenSER project were identified by us: - the need for a new reliable release - delayed or low-quality releases are affecting the credibility of the project - degradation of the project quality - testing, performance measurements and code quality control are overlooked - lack of the control, management and coordination of the project - critical project issues could not be handled or solved OpenSIPS Solutions: -------------------- OpenSIPS is a continuation of the OpenSER project - we have a moral obligation to develop and deliver the high quality and reliable software we envisioned when starting OpenSER. 1) a 6 months release cycle, but with no compromise for the technical issues 2) extensive testing and performance measurements before each major release 3) contributions are guaranteed to be answered and integrated (if accepted) in less than 1 month 4) code review, architecture control and integrated design for software quality 5) technical management with a strict developer hierarchy and developer cooperation to prevent deadlocks or un-availabilities 6) intellectual honesty by having decisions taken on value-based criteria and arguments. To find more about the project, see the official web page : http://www.opensips.org OpenSIPS 1.4.0 release: ----------------------- First major release of OpenSIPS is n**ow available. It incorporates new functionalities like * dialog profiling extension * local route for internally generated requests * dial plan / translation module * general NAT traversal module (for non-invite signalling) * peering and rate limit (traffic shaping) modules , enhancements like: * SRV load balancing (based on weights) * nonce re-usage verification (safer authentication) and bug fixing - code, specs and functionality bugs. The full Changelog is available on the web site: To get the OpenSIPS 1.4.0 version, see the download page at: http://www.opensips.org/index.php?n=Resources.Downloads Best regards, Bogdan

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[Kamailio-Users] 603 Declined
by Kaeman Chris 10 Aug '08

10 Aug '08

Hello, Does openser as soon as receives 603 Declined from the termination kills the call and not route it to the failover route? Is this correct? Thanks, KC --------------------------------- Unlimited freedom, unlimited storage. Get it now

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[Serusers] Gentilini, Paul is out of the office.
by PGentilini＠intellicare.com 09 Aug '08

09 Aug '08

I will be out of the office starting 08/08/2008 and will not return until 08/11/2008. Please contact the CSC at 888-899-4227 if you need immediate assistance.

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[Kamailio-Users] Compiling MediaProxy 2.0 on CentOS 5
by Andrew Yager 09 Aug '08

09 Aug '08

Hi, Apologies if you have already seen this on the OpenSIPS users list; it seems that some of the domain changes are a little confusing for the moment. I'm trying to compile MediaProxy 2.0 on CentOS 5 for use with Kamailio. I've installed libnetfilter_conntrack from RPMForge, but I am getting an error with compiling. ./setup.py build running build running build_py running build_ext building 'mediaproxy.interfaces.system._conntrack' extension gcc -pthread -fno-strict-aliasing -DNDEBUG -O2 -g -pipe -Wall -Wp,- D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer- size=4 -m64 -mtune=generic -D_GNU_SOURCE -fPIC -fPIC - DMODULE_VERSION=2.0.3 -DIPTABLES_VERSION="1.4.0" -Imediaproxy/ interfaces/system/libiptc/include -I/usr/include/python2.4 -c mediaproxy/interfaces/system/_conntrack.c -o build/temp.linux- x86_64-2.4/mediaproxy/interfaces/system/_conntrack.o mediaproxy/interfaces/system/_conntrack.c: In function ‘ForwardingRule_init’: mediaproxy/interfaces/system/_conntrack.c:223: error: ‘NFCT_SOPT_SETUP_REPLY’ undeclared (first use in this function) mediaproxy/interfaces/system/_conntrack.c:223: error: (Each undeclared identifier is reported only once mediaproxy/interfaces/system/_conntrack.c:223: error: for each function it appears in.) error: command 'gcc' failed with exit status 1 Current IPTABLES info: % iptables --version iptables v1.3.5 % lsmod | grep conntrack ip_conntrack_netbios_ns 36033 0 ip_conntrack 91237 1 ip_conntrack_netbios_ns nfnetlink 40457 1 ip_conntrack Is there a package I am missing, or am I simply fighting a loosing battle and need to use a different flavor of Linux to get it to work easily? Thanks, Andrew -- Andrew Yager, Managing Director (MACS BCompSc MCP) Real World Technology Solutions Pty Ltd ph: 1300 798 718 or (02) 9037 0500 fax: (02) 9037 0591 mob: 0405 152 568 http://www.rwts.com.au/ or http://www.stonebridgecomputing.com.au/

1 0

[Kamailio-Users] failure route issue
by Mouncif Benniane 08 Aug '08

08 Aug '08

I am using openser 1.3, I can't make the failure_route to work properly. here is my config: modparam("tm", "fr_timer", 2) # set fr_timer route{ if (is_method("INVITE")) { avp_delete("*"); if (avp_db_query("SELECT prefix,grp_id,priority,custid,did,ip,port,failure_set,failure_i p,failure_port FROM test_table WHERE prefix='$rU'","$avp(s:prefix);$avp(s:grp_id);$avp(s:priority);$avp(s:custid);$avp(s:did);$avp(s:ip);$avp(s:port );$avp(s:failure_set);$avp(s:failure_ip);$avp(s:failure_ port)")) { $ru = "sip:"+$rU + "@" + $avp(s:ip) + ":" + $avp(s:port); if ($avp(s:failure_set) == y ) { $avp(i:3) = 3; t_on_failure("1"); t_relay(); } } } else { xlog("L_NOTICE", "avp_db_query() returned no records or failed query\n"); }; } ### END OF ROUTING failure_route[1] { # status is 408 if openser session timer fires if (t_check_status("408")) { $ru = "sip:"+$rU + "@" + $avp(s:failure_ip) + ":" + $avp(s:failure_port); t_relay(); } } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; exit; } So I set ip field to no responsive one and failure_ip to responsive one, but still the invite does not fail to the last one (failure_ip) after the timeout. Am I doing something wrong? cheers!!

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[Kamailio-Users] Kamailio in orange (logo submission).
by Alex Balashov 08 Aug '08

08 Aug '08

I have a logo submission from someone who works with me at Evariste - of artistic inclinations. I don't know what the proper submission path for the logos is, so I'm posting it here. Sorry if that's annoying. -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599

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[Kamailio-Users] parameter 'ftag' and 'avp'
by Steven Wu 08 Aug '08

08 Aug '08

Hi all, What is meaning of ftag or avp parameter in Record-Route herder? SER inserted this Record-Route. As far as know, OpenSER is come from SER. Best regards, Steven Wu

2 1

[Kamailio-Users] comments to pointed openser issues
by Daniel-Constantin Mierla 08 Aug '08

08 Aug '08

Hello, these issues were pointed in the first place, my comments follow inline. 1) the need for a new reliable release - delayed or low-quality releases are affecting the credibility of the project - http://lists.kamailio.org/pipermail/users/2008-August/018999.html - the link above shows this one was a faster release than in past - http://www.kamailio.net/mos/view/KAMAILIO-v1.4.0-Released/ - the link above shows the efforts in creating a testing suite which was running on devel server during this development cycle - the link above shows no much new things in the core, but code cleanup, duplicity removal, source code documentation 2) degradation of the project quality - testing, performance measurements and code quality control are overlooked - presented before, testing suite was built and running, lot of code was reviewed and improved - who in the new project will do that and couldn't do it because was blocked in the old project 3) lack of the control, management and coordination of the project - critical project issues could not be handled or solved - I am not aware of mis-coordinations that could affect the project so badly - I am not aware of critical issues that could not be solved and there was no report of a critical deadlock - if there was a failure in management, all from the board share same fault Moreover, I believe there are facts in the past that openser took care of its quality. I repeat as I said that QA might not be enough no matter is done. If openser 1.3.x is the core component of a platform hosting about 2 millions of subscribers does not ensure some QA, then nothing else can be said, see: http://www.ilocus.com/2008/03/the_largest_voip_offering_base.html None above motivates such reactions as happened recently, but if does, then probably with some openess and discussions everything would have been better now. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

1 0

[Kamailio-Users] OpenSIPS vs. Kamailio
by Darren Sessions 08 Aug '08

08 Aug '08

I have to say, this is probably one of the harder decisions I've had to make in a very long time. I have to say the OpenSER / SER decision was much easier. While I obviously don't know all the details of what's gone on between Bogdan and Henning or whomever else, I just really have a hard time swallowing Hennings explanation of what happened and his, what seems to be, emotional reaction. More so than that, it seems like there were quite a few decisions made by the board without immediately letting the community know about it, even quite a while after these decisions were made. While I do not dispute Hennings technical ability, It is my opinion that his disrespectful attitude, his "shotgun" conduct, and his obvious lack of management ability is not befitting of someone on Kamailio project's board regardless of who is at fault here. Honestly, I'm a bit disappointed that Daniel hasn't said more as well. I met him at the fall VON in 2005 and he was indeed very pleasant and I enjoyed our conversation. All I've heard is things like 'not wanting to discuss any further', etc. - but the fact remains that wether it's a comfortable topic or not, there are quite a few of us that are using OpenSER in a commercial environment and these are things that we NEED to know in order to make an informed decision. In the end, all I know is now there are two projects and I have to choose which one. Based on what I've read in the change-logs, my direct experience with Bogden in a technical support role directly related to SER at a previous business - my company will be using OpenSIPS going forward. I have a very high regard for virtually all of the Kamailio / OpenSER group, and again, I am disappointed and absolutely hate that I've been forced into a position to have to make a decision like this. I will continue to be active on both sets of lists for the foreseeable future to monitor progress. Having said all that, I can't tell you how much I hope things can still be resolved and we can all have a unified project. I would implore the Kamailio board to do whatever it takes to bring Bogdan back into the group for the sake of the community. Hopefully something like that won't be just a dream. Thanks, - Darren _____________________________ dmsessions(a)gmail.com http://www.darrensessions.com http://www.linkedin.com/in/dsessions _____________________________

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[Kamailio-Users] releases history
by Daniel-Constantin Mierla 08 Aug '08

08 Aug '08

Hello, as there were accuses about a delayed release, I am showing the dates when a major release was out: June 14, 2005 - first openser release, just a rebuild of ser 0.9.x October 28, 2005 - openser 1.0.0 July 10, 2006 - openser 1.1.0 (after more than 8 months) March 12, 2007 - openser 1.2.0 (after more than 8 months) December 13, 2007 - openser 1.3.0 (after more than 9 months) August 7, 2008 - openser 1.4.0 (less than 8 months) So, maybe my memory is damaged, but you should be able to learn it from mailing lists and web. I do not buy that motivation about project quality degradation in release management. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

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[Kamailio-Users] openser board summary
by Daniel-Constantin Mierla 08 Aug '08

08 Aug '08

Hello guys, first I apologize for the situation we are in and for my silence in the last days as I tried to get the release out and not put gas on fire. The board looks like being now an evil thing, maybe it is our fault we haven't given more details about insights, but there were not many as ones could imagine and the decisions were presented to community. Here are the outlines of the discussions on the board, since September 2005, when first mail came there: 2005-09 DCM: decide release number => resulted in openser 1.0.0 DCM: business mailing lists => resulted in creation of the list 2005-10 AG: asks about making AG a trusted CA for default delivery of openser => self signed or something invalid to force users to change 2005-11 DCM: starts discussion about making an openser statute DCM: puts in dicussion the behavior of Harry on the mailing lists DCM: contacts Harry 2005-12 ALL: continuation of previous mail discussions 2006-03 DCM: plans for new release 2006-04 JH: discussion about presence implementation 2006-05 BIA: about project hosting: still sourceforge or move out ALL: Sourceforge problems discussions JH: presented some options for hosting 2006-06 AG: sip summer code JH: about 1.1.0 2006-07 JH: presence server AG: minutes if SIMPLE WG at IETF DCM: GPL issue with openssl and debian 2006-08 BIA: openser summit 2006-09,10 ALL: openser summit 2006-11 AG: history update in relation with old project BIA: TLS discussion 2006-12 ALL: history update 2007-01 KD: FOSDEM DCM: openser 1.1.1 BIA: openser at spring von 2007-02 BIA: openser donnations DCM: openser events 2007-04 BIA: Linuxtag 2007-09 DCM: summit 2007 2007-10 CS: retires from board BIA: proposes Henning => no decision as no rules for electing, makes that a point in todo BIA: openser 1.2.3 2007-11 DCM: openser business environment => hosting it within the website couldn't get a decision, posponed with alternative for dedicated site BIA: new developer Maxim Sobolyev 2008-03 BIA: on the 28th announces to board the issues with trademark, apparently some board member and other people were already aware AG: openser pages update 2008-04 ALL: agreed with rename - started the idea of a foundation to register trademark, takes on managing resources HW (not a member at that time but was included in the discussions related to TM) - proposes new developer Iouri 2008-05 BIA: on the 12th started the discussion about new name foreseeing 2 directions: something similar, to remember about the old project, or something totally different 2008-06 DCM: 9th: paypal report for last year => total donations to project's sourceforge account 52.05Euro (DCM Paypal account closed, the money still at DCM, proposed to use them for buying domains for new name) DCM: 9th: forwards a proposal received at end of April to team(a)openser.org from PacktPub to donate 2% or 5% (if we keep news on the site about the book) of openser book sales to the project. DCM asked for decision as there was no legal entity to deal with. (note: team(a)openser.org was created at the time openser forked in 2005, to get private feedback at that time, was used for the summit 2006 for contact, open for any post, includes the founders, about 10 good mails out of thousands spams per year) BIA: 9th: adds that in a discussion with PacktPub (not on team or board, as I haven;t seen such) proposed to give the 5% for openser devel course organized by him. The course was announced to the lists on the 4th of June. DCM: 9th: expressed surprise as he thought that was a different deal, made privately between BIA and PacktPub BIA: ask the board about donation, saying that actually the 5% donation was not really accounted ALL: There was no decision of what to do with the sponsorship as the board has no treasurer and we are working towards creating a foundation that can accept legally fundings. There was no ask for some amount of money for the devel course, no costs presented, no other details given. Nobody expressed a disagreement of giving funds for the devel course. BIA: 9th : replacement for Cesc as quick solution to deal with TM => suggests Henning Westerholt, all the other agree up to 17th DCM: asks Henning and then sends short announcement to community BIA: complains that he actually hasn't voted and the replacement was made in hurry (?!?!) DCM: presents that all other announcements were done more or less in the same fashion in the past, asking to propose improvements DCM: 24th - asks decision about local_route, board discusses and agrees to enclose in #ifdef HW: 25th - announces the result and asks BIA if he agrees, no response, he was away HW: after 1 day since asking BIA for agreement SVN is updated enclosing local_route in ifdefs BIA: 27th - disagrees with board decision, considering it was not the business of the board (sent to public as well) -- Note from DCM: opensips was registered at June 25, 2008, 17:30 according to http://sourceforge.net/projects/opensips/ Discussions with name proposals and foundation 2008-07 Others: discussions related to name and foundation HW: proposes a project renaming plan HW: on the 16th, vote for name: proposes Kamailio, majority reached on the 17th, last that agree (with a mention that he does not really like the name) BIA, on the 28th, after a call from HW where he mentioned he was away. AG: 17th - registers kamailio.org, ... AG: prposes unification of 1.4. and renaming => decided to announce first to allow time for renaming without pressure AG: 28th - announces renaming to public HW: some drafts of foundation if in Germany AG: details about alternative in Netherlands 2008-08 DCM: 4th - discovers news on voip-info.org that openser becomes opensips, Dan Pascu sends a mixed recipient message and some others see something is going on DCM: after short investigation and learning that behind opensips is bogdan decide to take some precaution actions and send details about them on board to decide if they still remain or roll back BIA: comes later on the mailing lists and announces the fork From then on most of the things were reflected on the public. No one was kicked out from the project, as developer, board member or simply list user, just some administrative rights were limited. Of course last three months have a lot of messages for the topics above. I hope I haven't missed something important and I ask other board members to correct or improve. With this one I hope things are a bit more clear about what board did so far, where were mistakes and wrong decisions. You judge! Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

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[Serusers] ser 2.1 - compilation error - conflicting types
by Tomasz Zieleniewski 08 Aug '08

08 Aug '08

Hi, When I try to compile ser from sources I get the following conflicting types errors: gcc -m64 -g -O9 -funroll-loops -Wcast-align -minline-all-stringops -falign-loops -ftree-vectorize -mtune=opteron -Wall -DNAME='"ser"' -DVERSION='"2.1.0-dev23-make"' -DARCH='"x86_64"' -DOS='linux_' -DOS_QUOTED='"linux"' -DCOMPILER='"gcc 4.3.1"' -D__CPU_x86_64 -D__OS_linux -DSER_VER=2001000 -DCFG_DIR='"/usr/local/etc/ser/"' -DMODS_DIR='"/usr/local/lib/ser/modules/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DUSE_DNS_CACHE -DUSE_DNS_FAILOVER -DUSE_DST_BLACKLIST -DUSE_NAPTR -DDBG_QM_MALLOC -DUSE_TLS -DTLS_HOOKS -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DCC_GCC_LIKE_ASM -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -DHAVE_TIMEGM -DHAVE_SCHED_SETSCHEDULER -DHAVE_EPOLL -DHAVE_SIGIO_RT -DSIGINFO64_WORKARROUND -DUSE_FUTEX -DHAVE_SELECT -c lock_ops.c -o lock_ops.o In file included from ut.h:53, from lock_ops.c:28: /usr/include/sys/types.h:46: error: conflicting types for 'loff_t' /usr/include/linux/types.h:30: error: previous declaration of 'loff_t' was here /usr/include/sys/types.h:62: error: conflicting types for 'dev_t' /usr/include/linux/types.h:13: error: previous declaration of 'dev_t' was here In file included from /usr/include/sys/types.h:133, from ut.h:53, from lock_ops.c:28: /usr/include/time.h:105: error: conflicting types for 'timer_t' /usr/include/linux/types.h:22: error: previous declaration of 'timer_t' was here In file included from ut.h:53, from lock_ops.c:28: /usr/include/sys/types.h:198: error: conflicting types for 'int64_t' /usr/include/linux/types.h:98: error: previous declaration of 'int64_t' was here /usr/include/sys/types.h:204: error: conflicting types for 'u_int64_t' /usr/include/linux/types.h:97: error: previous declaration of 'u_int64_t' was here In file included from /usr/include/sys/types.h:220, from ut.h:53, from lock_ops.c:28: /usr/include/sys/select.h:78: error: conflicting types for 'fd_set' /usr/include/linux/types.h:12: error: previous declaration of 'fd_set' was here In file included from ut.h:53, from lock_ops.c:28: /usr/include/sys/types.h:235: error: conflicting types for 'blkcnt_t' /usr/include/linux/types.h:124: error: previous declaration of 'blkcnt_t' was here make: *** [lock_ops.o] Błąd 1 Is there any fix for that? Regards Tomasz

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[Kamailio-Users] mailing lists issues
by Daniel-Constantin Mierla 07 Aug '08

07 Aug '08

Hello, I am not going in hot discussions here. We have to finish the release first. Due to DNS change of openser.org, without prior notice, we were facing about 1 hour down time with mailing lists. I apologize for that. Kamailio project should be reachable on two domains: - main one: kamailio.org (www.kamailio.org) - backup: kamailio.net (www.kamailio.net) The mailing lists domains are: - main one: lists.kamailio.org - backup: lists.kamailio.net If you have any problem, please contact me. Best regards, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

2 1

Re: [Kamailio-Users] carrierroute compile error
by Mouncif Benniane 07 Aug '08

07 Aug '08

That fixed it thanks On Thu, Aug 7, 2008 at 4:03 PM, TCB <tawandac(a)gmail.com> wrote: > you need libconfuse-deve ! > > > On Thu, Aug 7, 2008 at 9:38 PM, Mouncif Benniane <mouncifbb(a)gmail.com>wrote: > >> I did install libconfuse rpm for centos 5 but still same compile results >> >> [root@r sip-server]# rpm -qa | grep confuse >> *libconfuse-2.5-4.el5* >> [root@r sip-server]# >> >> On Thu, Aug 7, 2008 at 3:30 PM, Daniel-Constantin Mierla < >> miconda(a)gmail.com> wrote: >> >>> You need libconfuse, don't know exactly the name of the package on >>> centos. >>> >>> http://www.kamailio.net/docs/modules/1.3.x/carrierroute.html#AEN53 >>> >>> Cheers, >>> Daniel >>> >>> PS. Please note that the new mailing list address is >>> users(a)lists.kamailio.net >>> >>> >>> On 08/07/08 22:25, Mouncif Benniane wrote: >>> >>>> I am compiling version openser 1.3.2 on centos realease 5 >>>> >>>> getting compile error stooping at carrierroute module see below: >>>> >>>> >>>> make[1]: Entering directory >>>> `/usr/local/src/openser-1.3.2/sip-server/modules/carrierroute' >>>> make[1]: Leaving directory >>>> `/usr/local/src/openser-1.3.2/sip-server/modules/carrierroute' >>>> make[1]: Entering directory >>>> `/usr/local/src/openser-1.3.2/sip-server/modules/carrierroute' >>>> Compiling carrierroute.c >>>> Compiling carrier_tree.c >>>> Compiling load_data.c >>>> Compiling route_config.c >>>> route_config.c:39:21: error: confuse.h: No such file or directory >>>> route_config.c:50: error: expected declaration specifiers or â...â >>>> before âFILEâ >>>> route_config.c:52: error: expected â=â, â,â, â;â, âasmâ or >>>> â__attribute__â before â*â token >>>> route_config.c:63: error: expected â)â before â*â token >>>> route_config.c: In function âload_configâ: >>>> route_config.c:81: error: âcfg_tâ undeclared (first use in this >>>> function) >>>> route_config.c:81: error: (Each undeclared identifier is reported only >>>> once >>>> route_config.c:81: error: for each function it appears in.) >>>> route_config.c:81: error: âcfgâ undeclared (first use in this function) >>>> route_config.c:83: error: âdâ undeclared (first use in this function) >>>> route_config.c:83: error: âpâ undeclared (first use in this function) >>>> route_config.c:83: warning: left-hand operand of comma expression has no >>>> effect >>>> route_config.c:83: error: âtâ undeclared (first use in this function) >>>> route_config.c:83: warning: left-hand operand of comma expression has no >>>> effect >>>> route_config.c:93: warning: implicit declaration of function >>>> âparse_configâ >>>> route_config.c:104: warning: implicit declaration of function âcfg_sizeâ >>>> route_config.c:112: warning: implicit declaration of function >>>> âcfg_getnsecâ >>>> route_config.c:113: warning: implicit declaration of function >>>> âcfg_titleâ >>>> route_config.c:113: warning: assignment makes pointer from integer >>>> without a cast >>>> route_config.c:118: warning: assignment makes pointer from integer >>>> without a cast >>>> route_config.c:123: warning: implicit declaration of function >>>> âcfg_getintâ >>>> route_config.c:127: warning: assignment makes pointer from integer >>>> without a cast >>>> route_config.c:132: warning: implicit declaration of function >>>> âcfg_getfloatâ >>>> route_config.c:134: warning: implicit declaration of function >>>> âcfg_getstrâ >>>> route_config.c:134: warning: assignment makes pointer from integer >>>> without a cast >>>> route_config.c:135: warning: assignment makes pointer from integer >>>> without a cast >>>> route_config.c:137: warning: assignment makes pointer from integer >>>> without a cast >>>> route_config.c:145: warning: implicit declaration of function >>>> âcfg_getnintâ >>>> route_config.c:169: warning: implicit declaration of function âcfg_freeâ >>>> route_config.c: At top level: >>>> route_config.c:178: error: expected â=â, â,â, â;â, âasmâ or >>>> â__attribute__â before â*â token >>>> route_config.c: In function âsave_configâ: >>>> route_config.c:234: error: âFILEâ undeclared (first use in this >>>> function) >>>> route_config.c:234: error: âoutfileâ undeclared (first use in this >>>> function) >>>> route_config.c:241: warning: implicit declaration of function âfopenâ >>>> route_config.c:249: warning: implicit declaration of function âfprintfâ >>>> route_config.c:249: warning: incompatible implicit declaration of >>>> built-in function âfprintfâ >>>> route_config.c:251: error: too many arguments to function >>>> âsave_route_data_recursorâ >>>> route_config.c:257: warning: implicit declaration of function âfcloseâ >>>> route_config.c: At top level: >>>> route_config.c:274: error: expected declaration specifiers or â...â >>>> before âFILEâ >>>> route_config.c: In function âsave_route_data_recursorâ: >>>> route_config.c:280: warning: incompatible implicit declaration of >>>> built-in function âfprintfâ >>>> route_config.c:280: error: âoutfileâ undeclared (first use in this >>>> function) >>>> route_config.c:326: error: too many arguments to function >>>> âsave_route_data_recursorâ >>>> route_config.c: In function âbackup_configâ: >>>> route_config.c:335: error: âFILEâ undeclared (first use in this >>>> function) >>>> route_config.c:335: error: âfromâ undeclared (first use in this >>>> function) >>>> route_config.c:335: error: âtoâ undeclared (first use in this function) >>>> route_config.c:335: warning: left-hand operand of comma expression has >>>> no effect >>>> route_config.c:364: warning: implicit declaration of function âfeofâ >>>> route_config.c:365: warning: implicit declaration of function âfgetcâ >>>> route_config.c:366: warning: implicit declaration of function âferrorâ >>>> route_config.c:370: warning: implicit declaration of function âfputcâ >>>> route_config.c:377: error: âEOFâ undeclared (first use in this function) >>>> make[1]: *** [route_config.o] Error 1 >>>> make[1]: Leaving directory >>>> `/usr/local/src/openser-1.3.0/sip-server/modules/carrierroute' >>>> make: *** [modules] Error 2 >>>> >>>> >>>> >>>> >>>> any ideas? thank you >>>> ------------------------------------------------------------------------ >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users(a)lists.kamailio.org >>>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >>>> >>>> >>> >>> -- >>> Daniel-Constantin Mierla >>> http://www.asipto.com >>> >>> >> >> _______________________________________________ >> Users mailing list >> Users(a)lists.kamailio.org >> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >> >> > > > -- > TC >

1 0

[Kamailio-Users] Accounting / REFER question
by Zahid Mehmood 07 Aug '08

07 Aug '08

Hi All, I have a need to keep the openser 'IN' the loop for all transfered calls and to be able to generate proper accounting records (to mimic legacy PBX behavior). Any thoughts on how this can be achieved WITHOUT using SBC/ B2BUA? is it even possible? If off-system user calls X on openser and then X transfers the call to Y, we need to be able to generate two cdr records: A --> X start when A called, end when Y disconnected X --> Y start when X called Y, end when Y disconnected. A is connecting to X via a cisco gateway. Y can potentially be another off-system number. Thanks in Advance. -- Zahid

2 1

[Kamailio-Users] carrierroute compile error
by Mouncif Benniane 07 Aug '08

07 Aug '08

I am compiling version openser 1.3.2 on centos realease 5 getting compile error stooping at carrierroute module see below: make[1]: Entering directory `/usr/local/src/openser-1.3.2/sip-server/modules/carrierroute' make[1]: Leaving directory `/usr/local/src/openser-1.3.2/sip-server/modules/carrierroute' make[1]: Entering directory `/usr/local/src/openser-1.3.2/sip-server/modules/carrierroute' Compiling carrierroute.c Compiling carrier_tree.c Compiling load_data.c Compiling route_config.c route_config.c:39:21: error: confuse.h: No such file or directory route_config.c:50: error: expected declaration specifiers or â...â before âFILEâ route_config.c:52: error: expected â=â, â,â, â;â, âasmâ or â__attribute__â before â*â token route_config.c:63: error: expected â)â before â*â token route_config.c: In function âload_configâ: route_config.c:81: error: âcfg_tâ undeclared (first use in this function) route_config.c:81: error: (Each undeclared identifier is reported only once route_config.c:81: error: for each function it appears in.) route_config.c:81: error: âcfgâ undeclared (first use in this function) route_config.c:83: error: âdâ undeclared (first use in this function) route_config.c:83: error: âpâ undeclared (first use in this function) route_config.c:83: warning: left-hand operand of comma expression has no effect route_config.c:83: error: âtâ undeclared (first use in this function) route_config.c:83: warning: left-hand operand of comma expression has no effect route_config.c:93: warning: implicit declaration of function âparse_configâ route_config.c:104: warning: implicit declaration of function âcfg_sizeâ route_config.c:112: warning: implicit declaration of function âcfg_getnsecâ route_config.c:113: warning: implicit declaration of function âcfg_titleâ route_config.c:113: warning: assignment makes pointer from integer without a cast route_config.c:118: warning: assignment makes pointer from integer without a cast route_config.c:123: warning: implicit declaration of function âcfg_getintâ route_config.c:127: warning: assignment makes pointer from integer without a cast route_config.c:132: warning: implicit declaration of function âcfg_getfloatâ route_config.c:134: warning: implicit declaration of function âcfg_getstrâ route_config.c:134: warning: assignment makes pointer from integer without a cast route_config.c:135: warning: assignment makes pointer from integer without a cast route_config.c:137: warning: assignment makes pointer from integer without a cast route_config.c:145: warning: implicit declaration of function âcfg_getnintâ route_config.c:169: warning: implicit declaration of function âcfg_freeâ route_config.c: At top level: route_config.c:178: error: expected â=â, â,â, â;â, âasmâ or â__attribute__â before â*â token route_config.c: In function âsave_configâ: route_config.c:234: error: âFILEâ undeclared (first use in this function) route_config.c:234: error: âoutfileâ undeclared (first use in this function) route_config.c:241: warning: implicit declaration of function âfopenâ route_config.c:249: warning: implicit declaration of function âfprintfâ route_config.c:249: warning: incompatible implicit declaration of built-in function âfprintfâ route_config.c:251: error: too many arguments to function âsave_route_data_recursorâ route_config.c:257: warning: implicit declaration of function âfcloseâ route_config.c: At top level: route_config.c:274: error: expected declaration specifiers or â...â before âFILEâ route_config.c: In function âsave_route_data_recursorâ: route_config.c:280: warning: incompatible implicit declaration of built-in function âfprintfâ route_config.c:280: error: âoutfileâ undeclared (first use in this function) route_config.c:326: error: too many arguments to function âsave_route_data_recursorâ route_config.c: In function âbackup_configâ: route_config.c:335: error: âFILEâ undeclared (first use in this function) route_config.c:335: error: âfromâ undeclared (first use in this function) route_config.c:335: error: âtoâ undeclared (first use in this function) route_config.c:335: warning: left-hand operand of comma expression has no effect route_config.c:364: warning: implicit declaration of function âfeofâ route_config.c:365: warning: implicit declaration of function âfgetcâ route_config.c:366: warning: implicit declaration of function âferrorâ route_config.c:370: warning: implicit declaration of function âfputcâ route_config.c:377: error: âEOFâ undeclared (first use in this function) make[1]: *** [route_config.o] Error 1 make[1]: Leaving directory `/usr/local/src/openser-1.3.0/sip-server/modules/carrierroute' make: *** [modules] Error 2 any ideas? thank you

2 3

[Kamailio-Users] test
by Daniel-Constantin Mierla 07 Aug '08

07 Aug '08

please ignore -- Daniel-Constantin Mierla http://www.asipto.com

5 5

[Serusers] SIP 304 Temporary moved: sendto returning errno 1
by Vincent Pelletier 07 Aug '08

07 Aug '08

Hi. I built & installed ser-2.0.0-rc1 on a debian stable amd64. When I try to redirect a call (case where destination user redirects before answering), it gets an SIP error 477. Corresponding error from SER process running in foreground: 0(30330) ERROR: udp_send: sendto(sock,0x2aebd16fc038,1227,0,xx.xx.xx.xx:xxxx,16): Operation not permitted(1) 0(30330) msg_send: ERROR: udp_send failed 0(30330) ERROR: t_send_branch: sending request on branch 0 failed 0(30330) ERROR: sl_reply_error used: Unfortunately error on sending to next hop occurred (477/SL) Unfortunately, "1" (EPERM) is not documented in sento manpage, as far as I can see, so it's quite hard to track down that problem. Kernel version is: 2.6.24.5-grsec iptables is used, but putting netcat in place of SER and the client shows that kernel allows packets to reach client (and more simply, it is possible to do a direct call, it's just redirects which fail). It can be a grsec problem, I still have to check that part, but I wanted to take the chance of helpful feedback in the meantime. Regards, -- Vincent Pelletier

2 1

Re: [Kamailio-Users] www.openser.org
by Henning Westerholt 07 Aug '08

07 Aug '08

On Thursday 07 August 2008, Nuno Marques wrote: > Can someone tell me how to access the dokuwiki page? Is offline in both > projects (Kamailio and Opensips...). Hi Nuno, there is a wrong link in the page, should be fixed now. Cheers, Henning

1 0

[Kamailio-Users] www.openser.org
by Bogdan-Andrei Iancu 07 Aug '08

07 Aug '08

Staring from today, www.openser.org will point to www.opensips.org . The mainlists will continue on openser like they use to, but I invite everyone to move to the new mail lists created on opensips.org 1 - It is the time to move forward with the openser project, in a positive way, for a reliable future development and enhancement. 2 - Kamillio community used the access right they had on the www.openser.org server to redirect to kamillio without asking my consent even if the domain is a property of Voice System. Thanks, Bogdan

3 2

[Kamailio-Users] [Fwd: [Kamailio-Devel] branching 1.4.0 in one hour]
by Daniel-Constantin Mierla 07 Aug '08

07 Aug '08

I think this is also useful info for user community as well. Cheers, Daniel -------- Original Message -------- Subject: [Kamailio-Devel] branching 1.4.0 in one hour Date: Thu, 07 Aug 2008 18:09:35 +0300 From: Daniel-Constantin Mierla <miconda(a)gmail.com> Reply-To: miconda(a)gmail.com To: devel <devel(a)lists.kamailio.org> Hello, this is a last notification between branching the 1.4.0 release. If any developers is still doing last minute work, let us know. Branching is about one hour from now 18:00 CET. Of course, will still be possible to add afterwards, just that will need commits in two branches. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com _______________________________________________ Devel mailing list Devel(a)lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/devel -- Daniel-Constantin Mierla http://www.asipto.com

1 0

[Kamailio-Users] Is sh interface is needed?
by mahesh.peddi＠info-spectrum.com 07 Aug '08

07 Aug '08

Hi, I am working on OpenSER 1.3.2 version as presence server and i want to integrate with IMS. For this what are modifications need. Is sh interface is needed? And how to trace the presence information? Regards, Mahesh Peddi Infospectrum India Pvt. Ltd. Cell: +91 9765775176 IP-Phone Ext. - 764

2 1

[Kamailio-Users] REFER & accounting question
by Zahid Mehmood 07 Aug '08

07 Aug '08

Hi All, I have a need to keep the openser 'IN' the loop for all transfered calls and to be able to generate proper accounting records (to mimic legacy PBX behavior). Any thoughts on how this can be achieved WITHOUT using SBC/ B2BUA? is it even possible? If off-system user calls X on openser and then X transfers the call to Y, we need to be able to generate two cdr records: A --> X start when A called, end when Y disconnected X --> Y start when X called Y, end when Y disconnected. A is connecting to X via a cisco gateway. Y can potentially be another off-system number. Thanks in Advance. -- Zahid

1 0

[Kamailio-Users] For xcap-client to make run is xcap-server is needed?
by mahesh.peddi＠info-spectrum.com 07 Aug '08

07 Aug '08

Hi All, Presently i am working on OpenSER 1.3.2 version to make as presence server and i include the presence module. But i am getting some errors when i explored it seems xcap client was not include.Now i was include but my doubt is for xcap-client to make run is xcap-server is needed. If so where i can get it. Please help me on this. Thanks & Regards, Mahesh Regards, Mahesh Peddi Infospectrum India Pvt. Ltd. Cell: +91 9765775176 IP-Phone Ext. - 764

1 0

[Kamailio-Users] For xcap-client to make run is xcap-server is needed?
by mahesh.peddi＠info-spectrum.com 07 Aug '08

07 Aug '08

Hi All, Presently i am working on OpenSER 1.3.2 version to make as presence server and i include the presence module. But i am getting some errors when i explored it seems xcap client was not include.Now i was include but my doubt is for xcap-client to make run is xcap-server is needed. If so where i can get it. Please help me on this. Regards, Mahesh Peddi Infospectrum India Pvt. Ltd. Cell: +91 9765775176 IP-Phone Ext. - 764

1 0

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