- sr-users - kamailio.org

[OpenSER-Users] Recommendations for IM/Voice/Video client
by Adrian Georgescu 11 Apr '08

11 Apr '08

For those interested, you can monitor the progress of development for the multi party chat functionality based on MSRP here: http://www.msrprelay.org/ http://www.msrprelay.org/wiki/MSRPChatSoftwareDesign Adrian >>>>>>>>>>>> I've tryed Bria but in its web page it appears: IM Management - Send instant messages to groups - Use IM tabs for multiple IM conversations in one window - See IM alerts for incoming IM It's not as a IM conferencing in a server side, but maybe enought... Anyway, you can do a conference with IMC module and create contacts like "IM room 1", "IM room 2" in each client. Of course, for now there is no way to know which users are joining a conference. There are two RFC's about it: - SIP Event Package for Conference State http://tools.ietf.org/html/rfc4575 - SIP Call Control - Conferencing for User Agents http://tools.ietf.org/html/rfc4579 but AFAIK there are not yet client side implementation. -- Iñaki Baz Castillo

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[OpenSER-Users] (no subject)
by Hamed al-Suhli 10 Apr '08

10 Apr '08

Hi I'm new to openser and also Linux server install and configuration, I'm profissional at windows and its network, but need to install server for openser on local small lan. also will be used as backup for data. what is the easiest server to install? -- hamed suhli http://sourceforge.net/projects/tarmeez

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[OpenSER-Users] Recommendations for IM/Voice/Video client
by Robert McGilvray 10 Apr '08

10 Apr '08

This is a little off-topic, but figured it was a good place to ask. We're interested in deploying OpenSER coupled with a SIP client primarily used for IM, but also for voice and video conferencing. I'm looking for feedback and recommendations on clients from people that may do this already or have been involved with it. So far one of the better ones looks like Bria (a step up from Eyebeam) from CounterPath. One short coming though is the lack of IM conferencing. I could probably rig something up with IMC but our users would prefer to have a clicky GUI for it. This is for an enterprise and it'd be nice to be able to brand it. Thanks -Bob

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[OpenSER-Users] forwarding responses to early in TM
by Taisto Qvist 10 Apr '08

10 Apr '08

Hi folks, When I am doing forking I am getting a puzzling result concerning forwarding of responses, and I'm wondering how script needs tweaking. When recieving a 6xx on one branch, the TM-module automatically generates CANCEL to cancel the remaining branches(I wouldnt mind if this was configurable), but the puzzling thing is that it forwards the 603 _before_ it has terminated the remaining client transactions. (It sends cancel but it doesnt wait for final responses of the invites its cancelling, just forwards 6xx directly.) A similar scenario happends when I let Timer C pop. My TM-module will send a final response on the server-side at the same time as it starts to cancel the client side. But since one of the client txns just might result in a 2xx(INV) I cant forward the final response until ALL client txns are done, if I am going to follow the rfc properly. I thought the TM module would keep track of all client txns for "me", but it seems it isnt, so is this something I should "manually" handle in my script? Regards Taisto Qvist

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[Serusers] Automatic end of call
by inge 10 Apr '08

10 Apr '08

Hi all, In find some informations about an automatic end of call system. The aim is to cut the call after 2 hours. So it's not specificly link to a pre-paid system, but for all my ser user, I want to end the call after 2 hours. Is there a existing solution to do that ? Or maybe some idea to deal with this project ? Thank you !

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[OpenSER-Users] Accounting Time on dialogue SIP
by Marcio 10 Apr '08

10 Apr '08

Hi... Anybody could me help, I need account time within dialogue SIP( from reply of the Invite -> OK until the BYE-> finish dialogue) and to save for a SGBD (MySQL - That's autenthication's responsible)....... Does anyone know how...? Any hints.....?? thx... Marcio

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[OpenSER-Users] Can't register on OpenSER using SIPp
by Vladimir Romanov 10 Apr '08

10 Apr '08

Hello All! I can't register in OpenSER using SIPp. SIPp versions tested 2.0.1 Windows (TLS), Latest from SVN on linux X-Lite registers w/o any problems. In pcap file & messages you can see 2 registration attempts. first form x-lite (successful) and second from SIPp. This looks like OpenSER and SIPp have some differences in MD5 authentication. -- Vladimir Romanov

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[OpenSER-Users] Can't register on OpenSER using SIPp
by Vladimir Romanov 10 Apr '08

10 Apr '08

I can't register in OpenSER using SIPp. SIPp versions tested 2.0.1 Windows (TLS), Latest from SVN on linux X-Lite registers w/o any problems. In pcap file & messages you can see 2 regsitration attempts. first form x-lite (sucessfull) and second from SIPp. This looks like OpenSER and SIPp have some differences in MD5 authentification. -- Vladimir Romanov

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[OpenSER-Users] Sip server tutorial
by kriko 10 Apr '08

10 Apr '08

Hi! I was assigned to explore OpenSer, however I'm more green on this stuff. Is there a good basic documentation/tutorial on how sip servers and systems around works (basic stuff like routing)? -- (\__/) (='.'=) (")_(")

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[OpenSER-Users] Wants to reopen auth_diameter module
by Dilip 10 Apr '08

10 Apr '08

Hello Friends, There is a auth_diameter module in the openser,But now no one is working on that. I want to make it compatible with the opendiameter. So that we can implement AAA protocol in the complete manner. For that i want your help for the completion of this module. Regards, Dilip

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[OpenSER-Users] survey: Who is using serweb?
by Klaus Darilion 10 Apr '08

10 Apr '08

Which version do you use? Does it work at all with current version of openser (1.3)? PS: this is to find out if serweb supported can be removed from openser regards klaus

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[Serusers] good news for German ser fans: book
by Jiri Kuthan 09 Apr '08

09 Apr '08

I got today this book on my desk, courtesy of one of its authors. https://www.opensourcepress.de/index.php?26&backPID=178&tt_products=194 Whereas I was busy with a bunch of unrelated things, the book was circulating in the iptel office all day and has found its keen readers. Unfortunately still only in German lanugage. -jiri

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[Serusers] select for canonical username (alias)?
by Stefan Sayer 09 Apr '08

09 Apr '08

Hello, is there a select to get the canonical username? or is 'canonical username' now a deprecated concept? I know there is the uid, but the problem is, with 36 characters uid is too long for my purpose (user id for voicebox). thanks Stefan -- Stefan Sayer VoIP Services stefan.sayer(a)iptego.com www.iptego.com iptego GmbH Am Borsigturm 40 13507 Berlin Germany Amtsgericht Charlottenburg, HRB 101010 Geschaeftsfuehrer: Alexander Hoffmann

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[Serusers] about URI request
by liu york 09 Apr '08

09 Apr '08

hi i want to distribute tow alieas users use different pstn gw with same area code. i add below in my cfg.but it doesnt work.any advice for me ? many many thx. if (uri=~"^sip:0060.*@(my\.test\.com)"){ strip(2); prefix("21"); rewritehostport("1.2.3.4:5060"); # This is where PSTN Gateway installed route(1); return; }; if (uri=~"^sip:0060.*@(your\.test\.com)"){ strip(2); prefix("22"); rewritehostport("5.6.7.8:5060"); # This is where PSTN Gateway installed route(1); return; };

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[OpenSER-Users] Invalid cseq for aor - OpenSER stuck?
by Adrian A 09 Apr '08

09 Apr '08

I am running OpenSER 1.3.1 on a Debian 4.0 system. It seems that there's some sort of bug with OpenSER getting stuck and stops processing requests at least once a day. The logs only show the following line repeated every minute: ERROR:registrar:update_contacts: invalid cseq for aor <1302> At 1302 is a Cisco 7905 phone running the latest firmware. If I restart OpenSER then things get back to normal. Does anyone know what the problem might be?

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[Serusers] SER 0.9.7-pre3 and DNS SRV
by Steven C. Blair 09 Apr '08

09 Apr '08

I have a SER 0.9.7-pre3 system which is working fine except for call transfers. It turns out that when the NOTIFY is sent after the REFER method the proxy performs a queried for only A records not SRV. This results in an IP address being returned which is a non-SIP host. This is because the domain name is being used by different services some of which use A records for resolving the name and some, like SIP, are suppose to use SRV. How can I make this version of SER perform an SRV lookup instead of an A record lookup when initiating SIP messages? Thanks, Steve

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[OpenSER-Users] OpenSer and sip calls
by kriko 09 Apr '08

09 Apr '08

Hi! I'm trying to get openser working properly - it starts up, I can even regiter from softphone (linphone / twinkle) but I cannot make calls. OpenSer is running on 10.99.8.103:5060 (this is my machine, IP is bound to eth0), when I try to call to e.g.: sip:demovoice@10.99.8.48 I get some errors as in this log: http://files.myopera.com/kriko/code/openser.tar.gz Configuration is included too. Softphone clients are properly configured - it is the proxy that is causing me problems. Any kind of information is appreciated. -- kriko

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[OpenSER-Users] Thoughts on openSER + Asterisk scenarios
by Gabriel Ortiz Lour 09 Apr '08

09 Apr '08

Hi all, We are working here on a solution for a large scale VoIP system to cover our entire state, using both openSER and Asterisk. The project is on an initial stage, basically deciding architecture, so I would like to share our thoughts here in the mailing list and hopefully get some feedback to get starting on the right foot. We have our own private network (internet only on a further first step, no NAT), and we think about two possible scenarios: 1: The SIP accounts would register on the local Asterisk and those would talk to a SER core to connect them all, something like "Asterisk: Oh! this account is not mine so I'll send it to the core to take care for me" and the SER core would try to reach the other endpoint or return some error code. 2: All SIP accounts would be registered on the SER core, using the local asterisk servers as mediaservers and backup SIP proxies, where the SIP accounts would be registering on them only when there is some problem on the link with the SER core (and here they would have to have their own local PSTN gateways) This is the two basic ideas, so we would like to have any thoughts about the "pros and cons" of them. We have also planned many PSTN gateways that would be used for LCR (making all PSTN calls = local) for the entire system, that should be considered on a failure event. Thanks, Gabriel Ortiz PS.: Any help/thoughts are very welcome, as we will share all the experience and development on this project with the FOSS community. We are a government IT company from Paraná, Brasil aligned with the Free Software movement, and the aim is to document and package every aspect of this project.

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Re: [OpenSER-Users] [openser 1.3] loading module auth.so stop my server (bug?!)
by Pascal Maugeri 09 Apr '08

09 Apr '08

I'm still troubleshooting this issue and this morning I added more outputs in the source code of auth_mod.c. And I discovered: 1) when I set the configuration parameter "secret" of the module auth (modparam("auth", "secret", "secretphrase) ) in openser.cfg everything works fine ! 2) when I remove the configuration parameter "secret", the auth_mod.c calls the method generate_random_secret() as stated in the documentation. It enters in the loop for(i = 0; i < RAND_SECRET_LEN; i++) { LM_INFO("(3a)\n"); sec_rand[i] = 32 + (int)(95.0 * rand() / (RAND_MAX + 1.0)); } LM_INFO("(4)\n"); but never exit ! This is the reason why my server does not start and keeps blocked there. I modified the code above to print more logs, I can see in the logs the string (3a) once but I never (4): IMHO it is blocked somewhere in the first random computation. I'm running openser on Redhat Enterprise v4. Is there anything else I should configure in my environment ? Or I found a bug ? Regards, Pascal On Tue, Apr 1, 2008 at 12:45 PM, Pascal Maugeri <pascal.maugeri1(a)gmail.com> wrote: > Hi > > I'm still unable to have a working server 1.3 that loads the module > auth.so. > > It is weird because there is no error message in my log file also I set > the debug level to maximum. But after starting the openser service, it says > it is successful but there is no running process (pgrep openser does not > show anything, when it does if I remove the load auth.so)! > > How can I get more debugging info about the loading and initialization of > auth ? > > Can anybody share with me his or her configuration file ? I may miss > something in mine: > loadmodule "sl.so" > [...] > loadmodule "auth.so" > (I believe it is sufficient). > > Is there any known issue using Redhat Enterprise 4 ? > > Regards, > Pascal > > >

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Re: [OpenSER-Users] Specifying IP Address for SIP Responses
by Bogdan-Andrei Iancu 09 Apr '08

09 Apr '08

Hi Victor, if a UAC wants to have asymmetric signalling, just send from IP_A:PORT_A and put in top most via IP_B:PORT_B - via is used for routing back the reply. regards, Bogdan Victor Cartes wrote: > Thanks, Iñaki. But It is not what I really need. > > I need to send to an UAS a SIP Request, like OPTIONS, and tell it to > send the reply to a different transport address (IP Addr + Port) than > the source address of the request. > > This is, basically, in order to perform NAT discovering and typing > from the server. > > Thanks in advance, > > Víctor > > > > > 2008/4/7, Victor Cartes <ietf.sip(a)tivatechnologies.com > <mailto:ietf.sip@tivatechnologies.com>>: > > Actually not. What I need the UAS exactly to do is to send the SIP > reply to a different IP address than the source of the SIP Request > that it is replying to. Just how it sends the replies to the UDP > port specified in top most Via HF. > > > 2008/4/7, Bogdan-Andrei Iancu <bogdan(a)voice-system.ro > <mailto:bogdan@voice-system.ro>>: > > Hi Victor, > > Not sure I got your question correctly, but you mean like > something like "received" via param used to route back the > replies? > > Regards, > Bogdan > > Victor Cartes wrote: > > Hello to all, > > About SIP/UDP: Is there any way to specify the IP address > for SIP responses, in a similar fashion as port specified > on top most Via HF? > > Thanks in advance, > > Víctor > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)lists.openser.org <mailto:Users@lists.openser.org> > http://lists.openser.org/cgi-bin/mailman/listinfo/users > > > > >

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[OpenSER-Users] Sametime
by r007b17 09 Apr '08

09 Apr '08

Hi all, i have a lotus sametime server (SIP) and a Jabber server (XMPP). Is it possible messaging with each other using xmpp Module ? Thanks in advance. ____________________________________________________________________________________ You rock. That's why Blockbuster's offering you one month of Blockbuster Total Access, No Cost. http://tc.deals.yahoo.com/tc/blockbuster/text5.com

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Re: [OpenSER-Users] CPL and Postgres problems.
by Bogdan-Andrei Iancu 09 Apr '08

09 Apr '08

Hmm....that is something strange - probably there is something mixed with the postgres data types..... Could you open a bug report for further investigation? Regards, Bogdan Ola Karlsson wrote: > It was TEXT and I changed it to BYTEA and that seemed to do the trick. > > /Ola > > On Mon, Apr 7, 2008 at 11:57 AM, Bogdan-Andrei Iancu > <bogdan(a)voice-system.ro <mailto:bogdan@voice-system.ro>> wrote: > > So, you only changed the column type (for cpl_bin) from BLOB to > BYTEA ? > > Regards, > Bogdan > > Ola Karlsson wrote: > > Bogdan.. > > > > I changed the column type in the database to bytea for cgi-bin and > > stuff started working , seems the decode only happened for BLOB type > > not for text. > > > > When debugging I got a \ instead of 0x01 så figured something was > > wrong getting stuff out of the database. > > > > /Ola > > > > On Mon, Apr 7, 2008 at 10:50 AM, Bogdan-Andrei Iancu > > <bogdan(a)voice-system.ro <mailto:bogdan@voice-system.ro> > <mailto:bogdan@voice-system.ro <mailto:bogdan@voice-system.ro>>> > wrote: > > > > Hello Ola, > > > > That is strange as the len of the script is taken from the > > postgres driver (via openser API) - the returned columns must be > > BLOB type. Check the types of the cpl_* columns; otherwise I > > suspect a postgres driver (in openser) issue. > > > > Best regards, > > Bogdan > > > > Ola Karlsson wrote: > > > > Hi Users! > > > > Have a problem with CPL scripts and postgres. > > I've tried to make the simplest possible script for > debugging > > which resulted in the following in the 'cpl_xml' column. > > > > '<cpl>\012 <incoming>\012 <reject status="reject" > > reason="I reject anonymous calls"/>\012 > > </incoming>\012</cpl>\012\012' > > > > Loading it with openserctl that gets translated to this > in the > > cpl_bin field. > > > > > '\001\001\000\000\000\006\002\001\000\000\000\006\026\000\002\000\000\000\002[\000\001\000\031I > > reject anonymous calls\000\000' > > > > The reason I figure that postgres could have something to do > > with this is that I've got this working before(before being > > quite a while ago though. ) with MySQL. > > > > Turning on debugging gives this information. > > > > Mar 31 18:13:18 fuj /sbin/openser[601]: cpl-c B4script > > Mar 31 18:13:18 fuj /sbin/openser[601]: > > DBG:cpl-c:get_dest_user: trying to get user from new_uri > > Mar 31 18:13:18 fuj /sbin/openser[601]: > > DBG:cpl-c:get_dest_user: trying to get user from R_uri > > Mar 31 18:13:18 fuj /sbin/openser[601]: > > DBG:cpl-c:get_user_script: fetching script for user <olak> > > Mar 31 18:13:18 fuj /sbin/openser[601]: > > DBG:cpl-c:get_user_script: we got the script len=0 > > Mar 31 18:13:18 fuj /sbin/openser[601]: cpl-c After Script > > > > OpenSER says that the length of the script is 0 whatever > > script I load?. > > I changed that in the CPL-C module so we get the right > length > > of the compiled script but then it falls over on the sanity > > check for NODE_CPL = 001 , and if you skip that check it > > starts complaining about unknown nodes ( in my case 92 ) ... > > > > So my question is if anyone uses postgres and CPL, if so can > > you give me some pointers? > > Would be good to know if this happens to be a bug or an > > untested area in that case I have no problem fixing the > > problem just don't want to burn alot of time on something > > already fixed or me doing something stupid with > configuration. > > > > Regards > > /Ola Karlsson > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Users mailing list > > Users(a)lists.openser.org <mailto:Users@lists.openser.org> > <mailto:Users@lists.openser.org <mailto:Users@lists.openser.org>> > > http://lists.openser.org/cgi-bin/mailman/listinfo/users > > > > > > > > > > > _______________________________________________ > Users mailing list > Users(a)lists.openser.org <mailto:Users@lists.openser.org> > http://lists.openser.org/cgi-bin/mailman/listinfo/users > >

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[Serusers] Adding authentication through mySQL
by frq ser 09 Apr '08

09 Apr '08

Hi, I am trying to add user authentication through mySQL database to my SER (0.9.7-pre1). When I tried to run SER with the modifed ser.cfg file, it gave the error that it could not locate the 'mysql.so' module. Then I tried to 'make' the mysql module following the instructions given here: http://www.iptel.org/faq/why_does_ser_fail_to_start_with_an_error_load_modu… When I typed 'make modules=modules/mysql modules' at the SER source directory, it gives me the error message saying that it could not locate 'mysql.h' and 'errmsg.h' (complete log given at the end of this email). I've tried to locate these files on my computer but couldn't find them. Note that mySQL is working fine on its own. I need your suggestions as to how I should proceed. Thanks in advance for any help. Frq

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[Serusers] ser sip-t or sip-i
by mashudi 09 Apr '08

09 Apr '08

Dear all, is ser can handle sip-t or sip-i invite message ? have some one got experience with sip-t or sip-i using ser ?

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[OpenSER-Users] [Fwd: Re: Bug with OpenSER 1.3 Presence module : event handling.]
by Anca Vamanu 09 Apr '08

09 Apr '08

-------- Original Message -------- Subject: Re: [OpenSER-Users] Bug with OpenSER 1.3 Presence module : event handling. Date: Wed, 09 Apr 2008 11:20:04 +0300 From: Anca Vamanu <anca(a)voice-system.ro> To: Jeremy McNamara <jj(a)nufone.net> CC: Iñaki Baz Castillo <ibc(a)aliax.net>, users(a)openser.org References: <b769badb0710240220pe56ebe2mf617230fdb667586(a)mail.gmail.com> <200710241333.09880.ibc(a)aliax.net> <471F34B5.6040806(a)voice-system.ro> <47FBB0CE.4000909(a)nufone.net> Hi Jeremy, Firstly, no, the message: '"Missing or unsupported event header field value / ^Ievent= message-summary"' has nothing to do with presence. It is about another feature - notifications about whether the user has voice mail messages pending. Now, since presence does not work for you with xlite, I am wandering if you have clicked the option Show contact availability , or something like this, when editing the buddy details. It is at the end of the form. regards, Anca Vamanu Jeremy McNamara wrote: > Anca Vamanu wrote: > >> Hello, >> >> A major change in the presence structure occurred in openser 1.3. You >> now need to load two modules to have presence running: presence and >> presence_xml. >> Most probably you didn't load the presence_xml module and this is why >> the error message appears. >> >> > > Currently I am at Interop HotStage in Belmont, CA trying to deploy a > 'unified communication' situation to demonstrate for the Interop Vegas > show here in a few weeks. I am in charge of the OpenSER/Asterisk > components and presence is giving me a huge headache. > > Hopefully someone out there can show me what I am doing wrong. > > I am running Openser v1.3.1 from tarball on Ubuntu Gutsy, no xcap > server (haven't figured out if we really need one or not for our demo...) > > I have loaded both presence and presence_xml. As documented in the > reply (down below) if you notice there is a 'tab' character in the > event= presence, I too experience this, however mine is with > message-summary. > > I see the Subscribe/Publish messages coming into openser and they are > getting routed to the appropriate config block and processed as I have > seen documented, but every so often the "Missing or unsupported event > header field value / ^Ievent= message-summary" error pops up. > > Between two X-Lite softclients using "presence agent' method, we never > seem to get any presence notifications, I presume due to this error, > but I am unsure. > > All dependencies on both presence.so and presence_xml.so seem to be > met (verified with ldd) > > # ldd presence_xml.so > linux-gate.so.1 => (0xffffe000) > libxml2.so.2 => /usr/lib/libxml2.so.2 (0xb7e76000) > libc.so.6 => /lib/tls/i686/cmov/libc.so.6 (0xb7d2c000) > libdl.so.2 => /lib/tls/i686/cmov/libdl.so.2 (0xb7d27000) > libz.so.1 => /usr/lib/libz.so.1 (0xb7d12000) > libm.so.6 => /lib/tls/i686/cmov/libm.so.6 (0xb7ced000) > /lib/ld-linux.so.2 (0x80000000) > > (identical for presence.so) > > > Here are the relevant portions of my config: > > loadmodule "presence.so" > modparam("presence", "db_url", "mysql://XXX:XXX@localhost/openser") > modparam("presence", "max_expires", 60) > modparam("presence", "server_address", "sip:valid.ip.address:5060") > > loadmodule "presence_xml.so" > modparam("presence_xml", "db_url","mysql://XXX:XXX@localhost/openser") > modparam("presence_xml", "force_active", 1) > > ... > > if(is_method("PUBLISH|SUBSCRIBE")) { > route(2); > } > > ... > > route[2] { > xlog("L_INFO", "Dealing with $rm F:$fu T:$tu\n"); > sl_send_reply("100","trying"); > if (!t_newtran()) { > sl_reply_error(); > exit; > }; > > append_to_reply("Contact: <sip:valid.ip.address:5060>\r\n"); > if(is_method("PUBLISH")) { > if($hdr(Sender)!= NULL) { > handle_publish("$hdr(Sender)"); > } else { > handle_publish(""); > } > t_release(); > } else { > if( is_method("SUBSCRIBE")) { > handle_subscribe(); > t_release(); > } > } > exit; > } > > > > > If anyone wants to help me troubleshoot and make this work, I will > make sure you get credit for assisting us here at Interop. > > > > > Jeremy McNamara > > > >>> El Miércoles, 24 de Octubre de 2007, Antoine Fressancourt escribió: >>> >>> I've now installed SVN version and same erro occurs to me: >>> >>> Oct 24 13:32:37 [4653] ERROR:presence:handle_subscribe: Missing or >>> unsupported event header field value >>> Oct 24 13:32:37 [4653] ERROR:presence:handle_subscribe: >>> event= presence >>> >>> The client is Twinkle and presence worked well in 1.2.X. >> > > > > >

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[OpenSER-Users] Missing or unsupported event header field value - presence woes
by Jeremy McNamara 08 Apr '08

08 Apr '08

I am at Interop HotStage in Belmont, CA trying to deploy a 'unified communication' situation to demonstrate for the Interop Vegas show here in a few weeks. I am in charge of the OpenSER/Asterisk components and presence is giving me a huge headache. > Apr 8 11:34:12 openser openser[23798]: > ERROR:presence:handle_subscribe: Missing or unsupported event header > field value Apr 8 11:34:12 openser openser[23798]: > ERROR:presence:handle_subscribe: ^Ievent= message-summary Hopefully someone out there can help me figure out what is going wrong. I am running Openser v1.3.1 from tarball on Ubuntu Gutsy, no xcap server (haven't figured out if we really need one or not for our demo...) I have loaded both presence and presence_xml. If you notice there is what I believe to be a 'tab' character in the event= message-summary error response. I wonder if this tab is causing the issue. I see the Subscribe/Publish messages coming into openser and they are getting routed to the appropriate config block and processed as I have seen documented, but every so often the "Missing or unsupported event header field value and ^Ievent= message-summary" error pops up. Between two X-Lite softclients using 'presence agent' method, we never seem to get any presence notifications, I presume due to the error above, but I am unsure. All dependencies on both presence.so and presence_xml.so seem to be met (verified with ldd) # ldd presence_xml.so linux-gate.so.1 => (0xffffe000) libxml2.so.2 => /usr/lib/libxml2.so.2 (0xb7e76000) libc.so.6 => /lib/tls/i686/cmov/libc.so.6 (0xb7d2c000) libdl.so.2 => /lib/tls/i686/cmov/libdl.so.2 (0xb7d27000) libz.so.1 => /usr/lib/libz.so.1 (0xb7d12000) libm.so.6 => /lib/tls/i686/cmov/libm.so.6 (0xb7ced000) /lib/ld-linux.so.2 (0x80000000) (identical for presence.so) Here are the relevant portions of my config: loadmodule "presence.so" modparam("presence", "db_url", "mysql://XXX:XXX@localhost/openser") modparam("presence", "max_expires", 60) modparam("presence", "server_address", "sip:valid.ip.address:5060") loadmodule "presence_xml.so" modparam("presence_xml", "db_url","mysql://XXX:XXX@localhost/openser") modparam("presence_xml", "force_active", 1) ... if(is_method("PUBLISH|SUBSCRIBE")) { route(2); } ... route[2] { xlog("L_INFO", "Dealing with $rm F:$fu T:$tu\n"); sl_send_reply("100","trying"); if (!t_newtran()) { sl_reply_error(); exit; }; append_to_reply("Contact: <sip:valid.ip.address:5060>\r\n"); if(is_method("PUBLISH")) { if($hdr(Sender)!= NULL) { handle_publish("$hdr(Sender)"); } else { handle_publish(""); } t_release(); } else { if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } } exit; } If anyone wants to help me troubleshoot and make this work, I will make sure you get credit for assisting us here at Interop. Jeremy McNamara > > > >

2 6

[Serusers] call recording app
by Mihaly Zachar 08 Apr '08

08 Apr '08

Hi all, I got a job now, I need to build a call recording application, what stores the rtp streams of the calls, what routed to the application. There is an existing routing logic (special sip proxy) what decides if the call should be recorded or not. If the call needs to be recorded, this proxy will route the call to this recorder application. This app needs to send through the call to the next sip hop, but it must store the RTP streams on the fly. What we have now: - we have our own rtpproxy, we can modify it easily if we need to store the streams. - we have our own B2BUA what controls the rtpproxy I just wondering if we can use SER + SEMS for this, or should we build our own system ? What should be the most effective way ? Can it be done with SEMS easily ? Thanks for the suggestions, recommendations. Misi

2 1

[OpenSER-Users] downgrading or removing openser
by Robert McNaught 08 Apr '08

08 Apr '08

Hi, I have googled and found some posts relating to this, but no definitive answers: how can I downgrade from openser 1.2.x to openser 1.1.x - do I just recompile openser 1.1 and it overwrites whatever is in 1.2? there does not seem to be a make uninstall option for make in 1.2. Recompiling does not give any errors, but wont there be modules, etc that are still left over from 1.2.x? Can anyone offer some advice? Robert

2 1

[OpenSER-Users] Specifying IP Address for SIP Responses
by Victor Cartes 08 Apr '08

08 Apr '08

Hello to all, About SIP/UDP: Is there any way to specify the IP address for SIP responses, in a similar fashion as port specified on top most Via HF? Thanks in advance, Víctor

5 12

[OpenSER-Users] ERROR:presence:publ_notify: Could not send notify for presence
by gregor.sp 08 Apr '08

08 Apr '08

Hi all, I am using OpenXcap (0.99) and OpenSER (1.3), presence authorization is working and users receive NOTIFYs when users subscribe to presence. But after updating presence information with PUBLISH no NOTIFY are sent to clients. The debug messages the server shows when the server receives a PUBLISH are: ERROR:presence_xml:get_final_notify_body: while extracting the transformation node ERROR:presence_xml:pres_apply_auth: in function get_final_notify_body ERROR:presence:send_notify_request: in function apply_auth_nbody ERROR:presence:notify: sending Notify not successful ERROR:presence:publ_notify: Could not send notify for presence Somebody could help me with this? Thanks. Gregorio -- View this message in context: http://www.nabble.com/ERROR%3Apresence%3Apubl_notify%3A-Could-not-send-noti… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

3 2

[OpenSER-Users] Session not stop
by Sergey Babichev 08 Apr '08

08 Apr '08

Hi all! I have found out a problem at end of a call from voip. If a call the user finishes voip that session is not broken off. If from PSTN that session comes to the end successfully. PSTN Gateway -> Openser with mediaproxy -> VoIP This is log (grep mediaproxy): call from pstn to voip, voip user ended call Apr 8 12:45:40 ser1-2 mediaproxy[15711]: delete F8A89F32-47E11DD-A31EB2F1-2DDABEF8(a)x.x.x.x info= Apr 8 12:45:40 ser1-2 mediaproxy[15711]: session F8A89F32-47E11DD-A31EB2F1-2DDABEF8(a)x.x.x.x: 529/302/831 packets, 105720/58476/164196 bytes (caller/called/relayed) Apr 8 12:45:40 ser1-2 mediaproxy[15711]: session F8A89F32-47E11DD-A31EB2F1-2DDABEF8(a)x.x.x.x: ended. Apr 8 12:45:40 ser1-2 mediaproxy[15711]: execution time: 0.44 ms Apr 8 12:45:41 ser1-2 mediaproxy[15711]: delete F8A89F32-47E11DD-A31EB2F1-2DDABEF8(a)x.x.x.x info= Apr 8 12:45:41 ser1-2 mediaproxy[15711]: execution time: 0.15 ms Apr 8 12:45:42 ser1-2 mediaproxy[15711]: delete F8A89F32-47E11DD-A31EB2F1-2DDABEF8(a)x.x.x.x info= Apr 8 12:45:42 ser1-2 mediaproxy[15711]: execution time: 0.15 ms Apr 8 12:45:44 ser1-2 mediaproxy[15711]: delete F8A89F32-47E11DD-A31EB2F1-2DDABEF8(a)x.x.x.x info= Apr 8 12:45:44 ser1-2 mediaproxy[15711]: execution time: 0.15 ms Apr 8 12:45:48 ser1-2 mediaproxy[15711]: delete F8A89F32-47E11DD-A31EB2F1-2DDABEF8(a)x.x.x.x info= Apr 8 12:45:48 ser1-2 mediaproxy[15711]: execution time: 0.15 ms Apr 8 12:45:52 ser1-2 mediaproxy[15711]: delete F8A89F32-47E11DD-A31EB2F1-2DDABEF8(a)x.x.x.x info= Apr 8 12:45:52 ser1-2 mediaproxy[15711]: execution time: 0.15 ms Apr 8 12:45:56 ser1-2 mediaproxy[15711]: delete F8A89F32-47E11DD-A31EB2F1-2DDABEF8(a)x.x.x.x info= Apr 8 12:45:56 ser1-2 mediaproxy[15711]: execution time: 0.17 ms Apr 8 12:46:00 ser1-2 mediaproxy[15711]: delete F8A89F32-47E11DD-A31EB2F1-2DDABEF8(a)x.x.x.x info= Apr 8 12:46:00 ser1-2 mediaproxy[15711]: execution time: 0.15 ms Apr 8 12:46:04 ser1-2 mediaproxy[15711]: delete F8A89F32-47E11DD-A31EB2F1-2DDABEF8(a)x.x.x.x info= Apr 8 12:46:04 ser1-2 mediaproxy[15711]: execution time: 0.15 ms Apr 8 12:46:08 ser1-2 mediaproxy[15711]: delete F8A89F32-47E11DD-A31EB2F1-2DDABEF8(a)x.x.x.x info= Apr 8 12:46:08 ser1-2 mediaproxy[15711]: execution time: 0.15 ms Apr 8 12:46:30 ser1-2 mediaproxy[15711]: delete F8A89F32-47E11DD-A31EB2F1-2DDABEF8(a)x.x.x.x info= Apr 8 12:46:30 ser1-2 mediaproxy[15711]: execution time: 0.17 ms Apr 8 12:46:30 ser1-2 mediaproxy[15711]: delete F8A89F32-47E11DD-A31EB2F1-2DDABEF8(a)x.x.x.x info= Apr 8 12:46:30 ser1-2 mediaproxy[15711]: execution time: 0.16 ms Apr 8 12:46:31 ser1-2 mediaproxy[15711]: delete F8A89F32-47E11DD-A31EB2F1-2DDABEF8(a)x.x.x.x info= Apr 8 12:46:31 ser1-2 mediaproxy[15711]: execution time: 0.15 ms Apr 8 12:46:33 ser1-2 mediaproxy[15711]: delete F8A89F32-47E11DD-A31EB2F1-2DDABEF8(a)x.x.x.x info= Apr 8 12:46:33 ser1-2 mediaproxy[15711]: execution time: 0.16 ms Apr 8 12:46:37 ser1-2 mediaproxy[15711]: delete F8A89F32-47E11DD-A31EB2F1-2DDABEF8(a)x.x.x.x info= Apr 8 12:46:37 ser1-2 mediaproxy[15711]: execution time: 0.15 ms This is log (grep mediaproxy): call from pstn to voip, pstn user ended call Apr 8 12:53:26 ser1-2 mediaproxy[15711]: session 109131A1-48011DD-A33BB2F1-2DDABEF8(a)x.x.x.x: ended. Apr 8 12:53:26 ser1-2 mediaproxy[15711]: execution time: 0.40 ms What's problem maybe? -- Best regards, Sergey Babichev

1 0

[OpenSER-Users] SeasMessageEvent [ExpressMChannel[12]] - ERROR parsing a SEAS message event!!!
by bsb093 08 Apr '08

08 Apr '08

Hi all, I am trying to run OpenSER 1.2 with WeSIP. After the callee (shine_sip1) answer the call I get the following exception and the call hangs up after a BYE message from the caller to the calee. ExpressMessageChannel [ExpressMChannel[12]] - Done processing ACK sip:shine_sip1@192.168.2.101:5070 SIP/2.0 ExpressMessageChannel [ExpressMChannel[12]] - peerAddress = 192.168.2.100/5060 SeasMessageEvent [ExpressMChannel[12]] - ERROR parsing a SEAS message event!!! java.lang.StringIndexOutOfBoundsException: String index out of range: -243 at java.lang.String.checkBounds(Unknown Source) at java.lang.String.<init>(Unknown Source) at com.voztele.javax.sip.header.Via.<init>(Via.java:112) at com.voztele.javax.sip.header.ViaList.<init>(ViaList.java:43) at com.voztele.javax.sip.message.SIPMessage.createSIPHeader(SIPMessage.java:247) at com.voztele.javax.sip.message.SIPMessage.addSIPHeader(SIPMessage.java:230) at com.voztele.javax.sip.message.SIPMessage.<init>(SIPMessage.java:225) at com.voztele.javax.sip.message.SIPRequest.<init>(SIPRequest.java:154) at com.voztele.javax.sip.parser.StringMsgParser.parseSERMessage(StringMsgParser.java:355) at com.voztele.seas.SeasMessageEvent.getSIPMessage(SeasMessageEvent.java:136) at com.voztele.seas.SeasTransactionEvent.getSIPMessage(SeasTransactionEvent.java:70) at com.voztele.seas.ExpressMessageChannel.run(ExpressMessageChannel.java:301) at java.lang.Thread.run(Unknown Source) Exception in thread "ExpressMChannel[12]" java.lang.NullPointerException at com.voztele.seas.SeasMessageEvent.getSIPMessage(SeasMessageEvent.java:143) at com.voztele.seas.SeasTransactionEvent.getSIPMessage(SeasTransactionEvent.java:70) at com.voztele.seas.ExpressMessageChannel.run(ExpressMessageChannel.java:301) at java.lang.Thread.run(Unknown Source) SeasMessageEvent [SeasRouter] - New message received, from 192.168.2.100:5060 to 192.168.2.100:5060 over UDP Please advice what can cause this exception. Thanks in advance, Shine -- View this message in context: http://www.nabble.com/SeasMessageEvent----------ExpressMChannel-12-----ERRO… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

1 0

[OpenSER-Users] authentication using openser
by bhargav.b＠renovau.com 08 Apr '08

08 Apr '08

hi all, i am trying to do authentication using openser at the front end and diameter as the back end.i have installed openser and opendiameter and they are running fine independently.but when i send needed avp's from openser towards opendiameter,the opendiameter client is clsing the connection.i donot understand why this is happening.if any body have any idea or if anybody provide me the refference to any study material regarding the integration of openser and diameter client,it would really help me a lot. seeking ur help regards bhargav.b

1 0

[OpenSER-Users] tee with openser
by Robert McNaught 07 Apr '08

07 Apr '08

Hi, This is probably a simple one - I have tried googling, but am not getting the answer to this. I am trying to catch what is displayed on screen and put it into a text file using tee so I can look at it. I can do 'free -h | tee dump.txt' and dump.txt is populated with data I can do 'asterisk -cvvvv | tee dump.txt' and dump.txt is populated with data when i try 'openser | tee dump.txt' and dump.txt does not get populated with the screen info. I have changed permissions to 666 on the dump.txt file incase that was the issue. Am I doing some silly? I thought that tee only copied what was displayed on screen. Thanks Robert

2 1

[OpenSER-Users] openser devel guide
by Daniel-Constantin Mierla 07 Apr '08

07 Apr '08

Hello, I want to announce the availability of the first version for openser devel guide, online at: http://www.asipto.com/pub/openser-devel-guide/ Written together with Elena-Ramona Modroiu, the tutorial approaches the important components of openser: shared memory, locking, pseudo-variables, transformations, mi commands, statistics, db api, timer, module development a.s.o. Hopefully, the guide will help new developers to understand easier the internals of openser and contribute back new code or improvements to existing code. Has a set of examples and references to the code to be able to learn quicker how to use the APIs. The document is under polishing process towards clean docbook and pretty printing (pdf) format, then the sources will be included in the openser repository to be more accessible for improvements. It will be a continuing work-in-progress document, please report any error you find there. It is also mirrored at: http://www.openser.org/docs/openser-devel-guide/ Cheers, Daniel -- http://www.asipto.com

3 4

[OpenSER-Users] OpenSER Summit 2008 - Post facts
by Bogdan-Andrei Iancu 07 Apr '08

07 Apr '08

Hello everybody, The most important OpenSER event, community oriented, the OpenSER Summit 2008 marked a new point in OpenSER project history. Gathering more than 60 participants from both industry and community side, OpenSER Summit brought in to the lights major project people and industry representatives who help with the evolution of the project. Major topics that were discussed: A. how can OpenSER be used as a core component in industry proofed solutions and how/where to get the necessary knowledge B. what are the development directions that needs to be taken by the project - these were moderated open discussions. All this theoretical discussions were back-end up by the OpenSER Pavilion, were several companies showed the practical part of OpenSER software, via services, products and knowledge. All the material that were presented during the Summit are now available for download: A. via the summit agenda - http://www.openser.org/mos/view/OpenSER-Summit-2008/#agenda - here, by browsing the agenda, you can download a certain presentation; B. via the pub/ section of the web site - http://www.openser.org/pub/OpenSER-Summit-2008/ - gets all the presentations from here. Pictures and video captures will be soon available on the web site ;) Best regards, Bogdan

1 1

Re: [OpenSER-Users] CPL and Postgres problems.
by Bogdan-Andrei Iancu 07 Apr '08

07 Apr '08

So, you only changed the column type (for cpl_bin) from BLOB to BYTEA ? Regards, Bogdan Ola Karlsson wrote: > Bogdan.. > > I changed the column type in the database to bytea for cgi-bin and > stuff started working , seems the decode only happened for BLOB type > not for text. > > When debugging I got a \ instead of 0x01 så figured something was > wrong getting stuff out of the database. > > /Ola > > On Mon, Apr 7, 2008 at 10:50 AM, Bogdan-Andrei Iancu > <bogdan(a)voice-system.ro <mailto:bogdan@voice-system.ro>> wrote: > > Hello Ola, > > That is strange as the len of the script is taken from the > postgres driver (via openser API) - the returned columns must be > BLOB type. Check the types of the cpl_* columns; otherwise I > suspect a postgres driver (in openser) issue. > > Best regards, > Bogdan > > Ola Karlsson wrote: > > Hi Users! > > Have a problem with CPL scripts and postgres. > I've tried to make the simplest possible script for debugging > which resulted in the following in the 'cpl_xml' column. > > '<cpl>\012 <incoming>\012 <reject status="reject" > reason="I reject anonymous calls"/>\012 > </incoming>\012</cpl>\012\012' > > Loading it with openserctl that gets translated to this in the > cpl_bin field. > > '\001\001\000\000\000\006\002\001\000\000\000\006\026\000\002\000\000\000\002[\000\001\000\031I > reject anonymous calls\000\000' > > The reason I figure that postgres could have something to do > with this is that I've got this working before(before being > quite a while ago though. ) with MySQL. > > Turning on debugging gives this information. > > Mar 31 18:13:18 fuj /sbin/openser[601]: cpl-c B4script > Mar 31 18:13:18 fuj /sbin/openser[601]: > DBG:cpl-c:get_dest_user: trying to get user from new_uri > Mar 31 18:13:18 fuj /sbin/openser[601]: > DBG:cpl-c:get_dest_user: trying to get user from R_uri > Mar 31 18:13:18 fuj /sbin/openser[601]: > DBG:cpl-c:get_user_script: fetching script for user <olak> > Mar 31 18:13:18 fuj /sbin/openser[601]: > DBG:cpl-c:get_user_script: we got the script len=0 > Mar 31 18:13:18 fuj /sbin/openser[601]: cpl-c After Script > > OpenSER says that the length of the script is 0 whatever > script I load?. > I changed that in the CPL-C module so we get the right length > of the compiled script but then it falls over on the sanity > check for NODE_CPL = 001 , and if you skip that check it > starts complaining about unknown nodes ( in my case 92 ) ... > > So my question is if anyone uses postgres and CPL, if so can > you give me some pointers? > Would be good to know if this happens to be a bug or an > untested area in that case I have no problem fixing the > problem just don't want to burn alot of time on something > already fixed or me doing something stupid with configuration. > > Regards > /Ola Karlsson > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)lists.openser.org <mailto:Users@lists.openser.org> > http://lists.openser.org/cgi-bin/mailman/listinfo/users > > > >

2 1

[OpenSER-Users] sdp_mangle_ip()
by Travis Hegner 07 Apr '08

07 Apr '08

Hello All, I have briefly looked through the bug list but couldn't find anything related to my problem. I thought I would check here first to see if perhaps I have a configuration issue. I'm currently using the latest 1.3 version downloaded with svn. When attempting to call "sdp_mangle_ip()" I end up with a malformed sip packet. Upon inspection, I realized that the packet is correct except for the "Content-Length:" header, which seems to have the new (correct) content-length of the new packet written twice into the data stream. Here is what a snippet of the packet looks like: Content-Length: 156\r\n156\r\n Obviously, a correct packet should look like: Content-Length: 156\r\n Has anyone experienced this error before? Is there some parameter I should be setting for the mangler module, or another function I should call before this one? Should I roll up my sleeves and see if I can patch it myself? Any info is greatly appreciated... Travis

2 1

[OpenSER-Users] CPL and Postgres problems.
by Ola Karlsson 07 Apr '08

07 Apr '08

Hi Users! Have a problem with CPL scripts and postgres. I've tried to make the simplest possible script for debugging which resulted in the following in the 'cpl_xml' column. '<cpl>\012 <incoming>\012 <reject status="reject" reason="I reject anonymous calls"/>\012 </incoming>\012</cpl>\012\012' Loading it with openserctl that gets translated to this in the cpl_bin field. '\001\001\000\000\000\006\002\001\000\000\000\006\026\000\002\000\000\000\002[\000\001\000\031I reject anonymous calls\000\000' The reason I figure that postgres could have something to do with this is that I've got this working before(before being quite a while ago though. ) with MySQL. Turning on debugging gives this information. Mar 31 18:13:18 fuj /sbin/openser[601]: cpl-c B4script Mar 31 18:13:18 fuj /sbin/openser[601]: DBG:cpl-c:get_dest_user: trying to get user from new_uri Mar 31 18:13:18 fuj /sbin/openser[601]: DBG:cpl-c:get_dest_user: trying to get user from R_uri Mar 31 18:13:18 fuj /sbin/openser[601]: DBG:cpl-c:get_user_script: fetching script for user <olak> Mar 31 18:13:18 fuj /sbin/openser[601]: DBG:cpl-c:get_user_script: we got the script len=0 Mar 31 18:13:18 fuj /sbin/openser[601]: cpl-c After Script OpenSER says that the length of the script is 0 whatever script I load?. I changed that in the CPL-C module so we get the right length of the compiled script but then it falls over on the sanity check for NODE_CPL = 001 , and if you skip that check it starts complaining about unknown nodes ( in my case 92 ) ... So my question is if anyone uses postgres and CPL, if so can you give me some pointers? Would be good to know if this happens to be a bug or an untested area in that case I have no problem fixing the problem just don't want to burn alot of time on something already fixed or me doing something stupid with configuration. Regards /Ola Karlsson

2 1

[OpenSER-Users] OpenSER on top of OpenIMSCore
by Habib, Yeasin 07 Apr '08

07 Apr '08

Hi, I want to install OpenSER server on top of OpenIMSCore. I am using the new VMWare image provided by Vladimir Romanov. I have tried to install it from the source from svn. But up to installation I was successful. But when I try to run the mysql database script it's not working. I am trying to run the script /usr/local/etc/openser/openserctlrc, but it's not working anyway. Because it doesn't seems like a shell script. If anybody installs OpenSER on top of OpenIMSCore VMWare image then please help me. If anybody has a working image of OpenSER then it will be really grateful if I have the link or tutorial of that image. I am looking forward to hearing from all of you. Best regards Yeasin Habib

2 1

[OpenSER-Users] How to detect lack of response to SIP nat ping?
by Mike Fedyk 07 Apr '08

07 Apr '08

Hi, I'm trying to come up with some code in openser that will log when a SIP UA does not respond to a SIP NAT ping. The best I can think of so far is using the sip timers module, but it is based on the INVITE method instead OPTIONS that is the nat ping default. Does anyone have any ideas that can point me in the right direction? Thanks, Mike

2 1

[OpenSER-Users] dlg_end_dlg with extra headers
by Alex Massover 07 Apr '08

07 Apr '08

Hi! I'm trying to use dlg_end_dlg with extra headers. As far as I understood I can send fifo command like: [root@SER openser]# openserctl fifo dlg_end_dlg 123 456 'Header: here\r\n' 400 Too few or too many arguments It doesn't seem to accept third extra-header parameter. Am I doing something wrong? -- Best Regards, Alex Massover VoIP Development Manager Jajah Inc.

2 1

[Serusers] SIP Registration fail
by William Zhang 06 Apr '08

06 Apr '08

Hi All, I am new to SER. I might just made some simple mistakes. So be easy on me:) I appreciate any inputs. Basically I was able to install and run SER 2.0.0 RC1 and make call between SIP clients SJPhone 1.65 with a helloworld configuration. Now I am trying to add the Authentication and MySQL. Here are what I did so far and no luck to get it work:( 1. Generate the configuration files with configuration file buildsystem and the content of the file is in the bottom of this message. 2. Create the SER database in MySQL server by running the ser_mysql.sh create command 3. Set SIP_DOMAIN variable: export SIP_DOMAIN="unknown.org" 4. Add domain and user with SER_CTL 2.0 with following commands ser_ctl domain add unknown.org ser_ctl user add 777(a)unknown.org -p 1212 The ser_ctl domain show and user show return: unknown.org unknown.org --------w--c-s 777(a)unknown.org uri 777(a)unknown.org 777(a)unknown.org credentials username=777 realm=unknown.org password=1212 777(a)unknown.org attr datetime_created=2008-04-06 03:25:43 I believe this should set the user and domain correctly to the database. But I also got some weired output with other command like ser_ctl ps, ser_ctl version. Same error: ser_ctl: (97, 'Address family not supported by protocol'): error'> And sercmd also give me the follow errors:( ERROR: connect_unix_sock: connect(/tmp/ser_ctl): Connection refused [111] 5. Run the ser with error to console and no fork. 6. Configure the SJPhone with above registration information. The client can not register and got 401 Unauthorized response. Below are the SIP packets contents: 2008-04-06 04:49:29.843 UDP LOCAL->192.168.0.182:5060 REGISTER sip:unknown.org SIP/2.0 Via: SIP/2.0/UDP 192.168.0.179;branch=z9hG4bKc0a800b30000004947f8565900000d5f00000019;rport From: "unknown" <sip:777@unknown.org>;tag=4ff22665b92 To: <sip:777@unknown.org> Contact: <sip:777@192.168.0.179> Call-ID: C69D49CA6580482CAFB5D305213ACE0D0xc0a800b3 CSeq: 18 REGISTER Max-Forwards: 70 User-Agent: SJphone/1.65.377a (SJ Labs) Content-Length: 0 21:49:29 SIP.Network DEBUG 2008-04-06 04:49:29.843 UDP 192.168.0.182:5060->LOCAL SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.0.179;branch=z9hG4bKc0a800b30000004947f8565900000d5f00000019;rport=5060 From: "unknown" <sip:777@unknown.org>;tag=4ff22665b92 To: <sip:777@unknown.org>;tag=3594b9925f930dcc8b38a6cda152d6b5.f97a Call-ID: C69D49CA6580482CAFB5D305213ACE0D0xc0a800b3 CSeq: 18 REGISTER Server: Sip EXpress router (2.0.0-rc1 (i386/linux)) Content-Length: 0 Warning: 392 192.168.0.182:5060 "Noisy feedback tells: pid=14897 req_src_ip=192.168.0.179 req_src_port=5060 in_uri=sip:unknown.org out_uri=sip:unknown.org via_cnt==1" The contents of the ser.cfg file: # Debug level on logging debug=3 # memory debug log level memdbg=10 # memory statistics log level memlog=10 # Fork and create children (no=run single process for debugging purposes) fork=no # Log to stderr (only use this of fork=no) log_stderror=yes # Log to the defined syslog facility log_facility=LOG_LOCAL0 # IP address to listen to listen=192.168.0.182 # Port to listen on port=5060 # Aliases used by uri!=myself checks to identity locally destined messages alias=mytestser.org # Number of children to fork per port per protocol children=4 # Is this host multihomed? mhomed=no # Should dns resolving be used? dns=no # Should reverse dns resolving be used? rev_dns=no # Should reverse dns resolving be used? disable_core_dump=yes # *************Your local directives ***************** # *** Common # *** Specific # Load the stateless transaction module (sl_reply etc) loadmodule "/usr/local/lib/ser/modules/sl.so" # Load the transaction stateful module (t_relay() etc) loadmodule "/usr/local/lib/ser/modules/tm.so" # Load the record routing module. Needed for making sure messages go through our server loadmodule "/usr/local/lib/ser/modules/rr.so" # Load maxfwd for stopping loops loadmodule "/usr/local/lib/ser/modules/maxfwd.so" # Load the user location module to store and retrieve user locations loadmodule "/usr/local/lib/ser/modules/usrloc.so" # Load the registrar module to handle registrations loadmodule "/usr/local/lib/ser/modules/registrar.so" # Load the xlog module for advanced logging with parameters loadmodule "/usr/local/lib/ser/modules/xlog.so" # Load the ctl module for the control socket interface. loadmodule "/usr/local/lib/ser/modules/ctl.so" # Load the gflags, global flags module. loadmodule "/usr/local/lib/ser/modules/gflags.so" # Load textops used for searching/selecting text in messages and manipulating headers. loadmodule "/usr/local/lib/ser/modules/textops.so" # Load the sanity check module to do integrity checks on incoming messages. loadmodule "/usr/local/lib/ser/modules/sanity.so" # Load the mysql connector module. loadmodule "/usr/local/lib/ser/modules/mysql.so" # Load the auth module with all the core authentication functions. loadmodule "/usr/local/lib/ser/modules/auth.so" # Load the auth module with the database functions. loadmodule "/usr/local/lib/ser/modules/auth_db.so" # The uri_db module uses database to check the uri. loadmodule "/usr/local/lib/ser/modules/uri_db.so" # The avp_db module allows loading of user attributes from database. loadmodule "/usr/local/lib/ser/modules/avp_db.so" loadmodule "/usr/local/lib/ser/modules/avp.so" # Load domain module used to verify the local domains. loadmodule "/usr/local/lib/ser/modules/domain.so" # *** Your loaded modules # Flags set be various modules # We set database mode to 0 = do not save to database backend (keep in memory), 1 = write-through to database modparam("usrloc", "db_mode", 1) # ctl module params # by default ctl listens on unixs:/tmp/ser_ctl if no other address is # specified in modparams; this is also the default for sercmd modparam("ctl", "binrpc", "unixs:/tmp/ser_ctl") # listen on the "standard" fifo for backward compatibility modparam("ctl", "fifo", "fifo:/tmp/ser_fifo") # We turn on full lr (adds lr=on as uri parameter, not lr). # This should have higher compatibility with some user agents. modparam("rr", "enable_full_lr", 1) # Encrypt dialog cookies with this secret modparam("rr", "cookie_secret", "ChangeThisIfYouStoreSecretStuffInCookie") # The authentication related modules and which database to use. modparam("auth_db|usrloc|uri_db", "db_url", "mysql://ser:heslo@localhost/ser") # We can store some of our important attributes (inside the messages) avpflags dialog_cookie; # ************* Your local module parameters ***************** # *** Common # *** Specific # Main route, executed for all messages route { # ------------------------------------------------------------------------ # Sanity Check Section # ------------------------------------------------------------------------ sanity_check(); # Check whether we might be in a loop. if (!mf_process_maxfwd_header("10")) { sl_reply("483", "Too Many Hops"); exit; } if (msg:len > max_len) { sl_reply("513", "Message Overflow"); exit; } # Log all incoming messages to see start of processing if(is_gflag("29")){ xlog("L_NOTICE","INCOMING - %rm\nFrom: %is(%fu)\nTo: %ru\nCall-ID: %ci\n"); } # ------------------------------------------------------------------------ # Handling messages with Route headers (aka loose routing) # ------------------------------------------------------------------------ if (loose_route()) { if(is_gflag("31")){ xlog("L_NOTICE"," loose routed - %rm\nFrom: %is(%fu)\nTo: %ru\nCall-ID: %ci\n"); } route(RELAY); exit; } # ------------------------------------------------------------------------ # Record Route Section # ------------------------------------------------------------------------ else if (method!="REGISTER") { record_route(); }; # If the ACK has more hops before reaching the user agent, it will have a Route header # and be forwarded above. If next hop is a user agent, the Request URI will be the ip:port # found in Contact of the original OK from the user agent, but will not be loose routed. # Just relay the ACK to the user agent. if (method=="ACK") { if(is_gflag("31")){ xlog("L_NOTICE"," ACK relayed using ruri - %rm\nFrom: %is(%fu)\nTo: %ru\nCall-ID: %ci\n"); } route(RELAY); exit; } # check if the caller is from a local domain (f=from, d=domain, $f will be set) lookup_domain("$fd", "@from.uri.host"); # check if the callee is at a local domain (t=to, d=domain, $t will be set) # If we have no To tag (dialog creating messages), we use request uri, but # but if this is an in-dialog message (like BYE with last hop before UA here) # we use the To header. if((a)to.tag=="") { lookup_domain("$td", "@ruri.host"); } else { lookup_domain("$td", "@to.uri.host"); } # we dont know the domain of the caller and also not # the domain of the callee -> someone uses our proxy as # a relay if (!$t.did && !$f.did) { sl_reply("403", "Relaying Forbidden"); exit; } # ------------------------------------------------------------------------ # Message Type Processing Section # NOTE!!! It is assumed that all messages reaching this point are destined # for the local server (REGISTER) or for UAs registered locally. (see nonlocal.m4) # ------------------------------------------------------------------------ if (method=="REGISTER") { route(REGISTER); exit; } # Authenticate route(AUTHENTICATE); # All local, non-ACKs, non-REGISTER are assumed to be part of a potential # new dialog (standalone MESSAGE and NOTIFYs without dialog-creation will ex. also reach here) route(NEWDIALOG); # By now, the request uri should be correct, relay if(is_gflag("29")){ xlog("L_NOTICE","No route matched for the request\nCall-Id: %ci\n"); }; sl_reply("404", "No route matched"); } # end main route route[RELAY] { # Log all outgoing messages if(is_gflag("29")){ xlog("L_NOTICE","OUTGOING - %rm\nFrom: %is(%fu)\nTo: %ru\nCall-ID: %ci\n"); } # ------------------------------------------------------------------------ # Default Message Handler # ------------------------------------------------------------------------ # t_relay will act based on the current request URI, as well as any # overruling received parameters (i.e. for NATed user agents) # UDP to TCP and vice versa works as well. if (!t_relay()) { sl_reply_error(); }; exit; } route[REGISTER] { # ------------------------------------------------------------------------ # REGISTER Message Handler # ------------------------------------------------------------------------ # From lookup_domain, $t.did should be set if we found the domain as ours. if (!$t.did) { if(is_gflag("29")){ xlog("L_NOTICE","%rm From <%fu> Attempted REGISTER from non-local domain\nCall-ID: %ci\n"); }; sl_reply("403", "Attempting to register with unknown domain"); exit; } # We check that Authorization header in the message contains valid credentials found in the # table with the same name. if (!www_authenticate("$fd.digest_realm", "credentials")) { # $? represents the return error code from www_authenticate() if ($? == -2) { if(is_gflag("31")){ xlog("L_NOTICE"," REGISTER - %rm From <%fu> 500 Internal Server Error\nCall-ID: %ci\n"); }; sl_reply("500", "Internal Server Error"); } else if ($? == -3) { if(is_gflag("31")){ xlog("L_NOTICE"," REGISTER - %rm From <%fu> 400 Bad Request\n"); }; sl_reply("400", "Bad Request"); } else { # After www_authenticate(), $digest_challenge should be set if no credentials were found. # We append the challenge to reply and send 401 Unauthorized. if ($digest_challenge) { append_to_reply("%$digest_challenge"); } sl_reply("401", "Unauthorized"); } # We didn't have a valid set of credentials, so we exit. exit; } # www_authenticate set $uid, so let's store it. $authuid=$uid; # Check if the authenticated user is the same as the target user. # We use the uri in the To field to lookup the user's uid. If none is found, the To user is unknown. if (!lookup_user("$tu.uid", "@to.uri")) { sl_reply("404", "Trying to register unknon user (To)"); exit; } # $f.uid set by www_authenticate() should match the uid we should looked up if ($authuid != $tu.uid) { sl_reply("403", "Authentication and To-Header mismatch"); exit; } # We use the uri in From to look up the uid. if (!lookup_user("$fu.uid", "@from.uri")) { sl_reply("404", "Expect a known user in From"); exit; } # The looked up $fu.uid is checked against the Authentication user if ($authuid != $fu.uid) { sl_reply("403", "Authentication and From-Header mismatch"); exit; } if (!save_contacts("location")) { sl_reply("400", "Invalid REGISTER Request"); exit; }; exit; } # End route[REGISTER] route[AUTHENTICATE] { # RFC3261 says you are not allowed to challenge these methods if (method=="CANCEL" || method=="ACK") { return; } # We check that Authorization header in the message contains valid credentials found in the # table with the same name. if (!proxy_authenticate("$fd.digest_realm", "credentials")) { # $? represents the return error code from proxy_authenticate() if ($? == -2) { if(is_gflag("31")){ xlog("L_NOTICE"," AUTHENTICATE - %rm From <%fu> 500 Internal Server Error\nCall-ID: %ci\n"); }; sl_reply("500", "Internal Server Error"); } else if ($? == -3) { if(is_gflag("31")){ xlog("L_NOTICE"," AUTHENTICATE - %rm From <%fu> 400 Bad Request\n"); }; sl_reply("400", "Bad Request"); } else { # After www_authenticate(), $digest_challenge should be set if no credentials were found. # We append the challenge to reply and send 407 Proxy Authentication Required. if ($digest_challenge) { append_to_reply("%$digest_challenge"); } sl_reply("407", "Proxy Authentication Required"); } # We didn't have a valid set of credentials, so we exit. exit; } # proxy_authenticate set $uid, so let's store it. $authuid=$uid; # Use the uri in the From header and look up the uid. If not found, delete the uri attribute. if (!lookup_user("$fu.uid", "@from.uri")) { del_attr("$uid"); } # Check the found uid against the authenticated uid. if ($fu.uid != $authuid) { sl_reply("403", "Fake Identity"); exit; } # load the user attributes (preferences) of the caller. $fu.yourattr should now be set if set in the # user preferences table. load_attrs("$fu", "$f.uid"); } route[NEWDIALOG] { # If we don't know the user, it is not for a local user. if (!lookup_user("$tu.uid", "@ruri")) { # User is not known and request uri does not match PSTN dialplan. return; } if (!lookup_contacts("location")) { if(is_gflag("30")){ xlog("L_NOTICE"," NEWDIALOG - 480 User Offline\nCall-Id: %ci\n"); }; sl_reply("480", "User Offline"); exit; } # Relay it to the location(s) of the found user route(RELAY); exit; } # End route[NEWDIALOG]

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[OpenSER-Users] Res: Openser with freeradius not accounting BYE method
by Flavio Goncalves 05 Apr '08

05 Apr '08

Set the flag 2 and 3 in the loose route for BYE requests. if (loose_route()) { if (is_method("BYE")) { setflag(2); setflag(3); }; }; ----- Mensagem original ---- De: MagoDaViola <magodaviola(a)ig.com.br> Para: users(a)lists.openser.org Enviadas: Sábado, 5 de Abril de 2008 16:50:56 Assunto: [OpenSER-Users] Openser with freeradius not accounting BYE method I´m has a problem that my openser sometimes not accounting bye method and this cause a call without pay. See the openser.cfg: # # $Id: ser.cfg,v 1.25.2.1 2005/02/18 14:30:44 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=4 # debug level (cmd line: -dddddddddd) ##-- # # Uncomment these lines to daemonize fork=yes log_stderror=no # (cmd line: -E) ##-- # # Uncomment these lines to enter debugging mode #fork=no #log_stderror=yes ##-- alias=sip.processavoip.net listen=201.32.70.20 #syn_branch=1 memlog=3 sip_warning=1 server_signature=yes reply_to_via=0 mhomed=0 disable_dns_blacklist=yes unix_sock="/tmp/openser.sock" check_via=0 # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 mpath="/usr/local/lib/openser/modules/" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "mysql.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "auth.so" loadmodule "auth_radius.so" loadmodule "uri.so" loadmodule "domain.so" loadmodule "permissions.so" loadmodule "mediaproxy.so" loadmodule "nathelper.so" loadmodule "acc.so" loadmodule "xlog.so" loadmodule "dbtext.so" loadmodule "uri_db.so" loadmodule "mi_fifo.so" loadmodule "group.so" # ----------------- setting module-specific parameters --------------- # -- mi-fifo params -- modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") # -- usrloc params -- modparam("usrloc", "db_mode", 2) modparam("usrloc", "nat_bflag", 6) modparam("usrloc", "domain_column", "fk_domains") modparam("usrloc", "user_column", "fk_accounts") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) ## -- database params -- modparam("acc|usrloc|group|domain|uri_db|permissions", "db_url", "mysql://root:master@localhost/openser") # -- radius params -- modparam("acc|auth_radius", "radius_config", "/etc/radiusclient-ng/radiusclient.conf") # -- radius permissions -- modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") # -- group params -- modparam("group", "table", "accounts_groups") modparam("group", "user_column", "fk_accounts") modparam("group", "domain_column", "fk_domains") modparam("group", "group_column", "pk_accounts_groups") modparam("group", "use_domain", 0) ## -- acc params -- modparam("acc", "radius_missed_flag", 3) modparam("acc", "radius_flag", 2) #modparam("acc", "radius_extra", "via=$hdr(Via[*]); email=$avp(s:email)") modparam("acc", "log_level", 2) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;dst_user=$rU;dst_domain=$rd") modparam("acc", "radius_extra", "User-Name=$fU;Suffix=$fd;Sip-Translated-Request-URI=$ru") modparam("acc", "service_type", 15) modparam("acc", "failed_transaction_flag", 1) modparam("acc", "report_ack", 1) modparam("acc", "report_cancels", 1) modparam("acc", "log_missed_flag", 2) modparam("acc", "log_flag", 1) ## -- acc params without radius -- #modparam("acc", "db_missed_flag", 3) #modparam("acc", "db_flag", 2) #modparam("acc", "db_table_acc", "call_history") #modparam("acc", "db_table_missed_calls", "call_history_missed") # -- uri_db params -- modparam("uri_db", "subscriber_table", "accounts") modparam("uri_db", "subscriber_domain_column", "fk_domains") modparam("uri_db", "subscriber_user_column", "pk_accounts") # -- domain params -- modparam("domain", "domain_table", "domains") modparam("domain", "domain_col", "pk_domains") # -- nathelper, mediaproxy nd registrar params -- modparam("nathelper", "sipping_bflag", 7) modparam("nathelper", "sipping_from", "sip:pinger@openser.org") modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind NAT modparam("registrar|nathelper", "received_avp", "$avp(i:42)") # to configure nat at openser 1.2, 1.3 or 1.4 modparam("mediaproxy", "mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy", "natping_interval", 30) #modparam("mediaproxy", "sip_asymmetrics", "/etc/openser/sip-clients") #modparam("mediaproxy", "rtp_asymmetrics", "/etc/openser/rtp-clients") #modparam("registrar", "received_avp", "$avp(s:rcv)") # to configure nat at openser 1.2, 1.3 or 1.4 # -- tm params -- # set time for which ser will be waiting for a final response; # fr_inv_timer sets value for INVITE transactions, fr_timer # for all others #modparam("tm", "fr_inv_timer", 15 ) #Tempo maximo do ring antes de atender #modparam("tm", "fr_timer", 1) #modparam("tm", "wt_timer", 1) # ------------------------- request routing logic ------------------- # main routing logic route { # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Muitos Hops"); log(1, "486: Muitos Hops\n"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message to big"); log(1, "513: Message too bi \n"); exit; }; ##-- # Eh feita a gravacao de todas as mensagens para se certificar que todas as mensagens # passem pelo proxy o que eh importante particularmente se as entidades estao usando # um protocolo de transporte diferente ##-- if (method!="REGISTER") { record_route(); }; route(6); ##-- # Secao de finalizacao da chamada ##-- if (method=="BYE" || method=="CANCEL") { end_media_session(); }; ##-- # mensagens subsequentes dentro de um dialogo deveriam tomar um caminho pre-determinado # pela gravacao das rotas (record route) ##-- if (loose_route()) { # marca a logica de roteamento no pedido append_hf("P-hint: rr_enforced\r\n"); if (has_totag() && (method=="INVITE" || method=="ACK")) { xlog("L_ERR", "Main Route: Fixing SDP to $fu $rm (loose route)\n"); if (client_nat_test("3") || isflagset(5)) { setbflag(6); use_media_proxy(); }; }; xlog("L_ERR", "Main Route: Loose Route relaying\n"); route(1); exit; }; if (!uri==myself) { # marca a logica de roteamento no pedido append_hf("P-hint: rr_enforced\r\n"); xlog("L_ERR", "Main Route: Relaing mesage to proxy destination \n"); route(4); route(1); }; ##-- # Verifica baseado na tabela de dominios se a parte host da R-URI comtem um dos dominios # quais o proxy eh responsavel ##-- if (!is_uri_host_local()) { if (is_from_local() || allow_trusted()) { route(4); route(1); } else { sl_send_reply("403", "Forbinden..."); } exit; }; ##-- # A rotina abixo eh chamada para marcar o registro com NAT ##-- if (method=="CANCEL") { route(1); exit; } else if (method=="REGISTER") { route(2); exit; } else if (method=="INVITE") { route(3); exit; }; ##-- # Verifica se existem aliases para este endereco e se eles nao pertencem # a teste dominio encaminhamos para frente ##-- lookup("aliases"); if (uri!=myself) { xlog("L_ERR", "Main Route: aliases relayin to other domain proxy\n"); route(4); route(1); exit; }; ##-- # Verifica se encontra o usuario na base de localizacao ##-- if (!lookup("location")) { sl_send_reply("404", "Usuario nao encontrado"); exit; }; route(1); exit; } route[1] { ##-- # Envie isto agora; use "statefull fowarding" pois funciona de forma confiavel de UDP para TCP # Ativa a contabilizacao das chamadas ##-- xlog("L_ERR", "Route[1]: Relaying message\r\n"); setflag(2); ##-- # Ativa a contabilizacao das chamadas perdidas ##-- setflag(3); t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; } route[2] { ##-- # Tratamento padrao para as mensagens de Register ##-- sl_send_reply("100", "Registrando...."); xlog("L_ERR", "Route[2]: Trying Auth: $fu $si\n"); if (isflagset(5) && client_nat_test("7")) { xlog("L_ERR", "Route[2]: Client behind NAT - flag(6)\n"); setbflag(6); fix_nated_register(); force_rport(); # se quiser usar natping options descomente a linha abaixo # setbflag(7); }; if (!radius_www_authorize("")) { www_challenge("", "0"); xlog("L_ERR", "Route[2]: Client Unauthorized: $fu $si\r\n"); sl_send_reply("401", "Unauthorized"); exit; }; if (!check_to()) { xlog("L_ERR", "Route[2]: Client Unauthorized: $fu $si\r\n"); sl_send_reply("401", "Unauthorized"); exit; } else { xlog("L_ERR", "Route[2]: Authenticate: $fu $si\r\n"); }; consume_credentials(); if (!save("location")) { xlog("L_ERR", "Route[2]: Can't save register location\r\n"); sl_reply_error(); }; } route[3] { ##-- # Tratamento padrao para as mensagens de INVITE ##-- xlog("L_ERR", "Route[3]: INVITE: $fu -> $tu $ci\r\n"); # setflag(1); if (!allow_trusted()) { if (!radius_proxy_authorize("")) { proxy_challenge("", "0"); # Realm will be autogenerated exit; } else if (!check_from()) { sl_send_reply("403", "Usue from=ID"); xlog("L_ERR", "Route[3]: Error 403 $fu -> $tu\r\n"); exit; }; } consume_credentials(); ##-- # Verifica se existem aliases para este endereco e se eles nao pertencem # a teste dominio encaminhamos para frente ##-- lookup("aliases"); if (uri!=myself) { xlog("L_ERR", "Main Route: aliases relayin to other domain proxy\n"); route(4); route(1); }; ##-- # VoIP Local ##-- if (uri=~"sip:[1-9][0-9]+@.*") { xlog("L_NOTICE", "OpenSER: Ligacao VoIP local: $fU -> $tu\n"); } else { ##-- # Nacional ##-- if (uri=~"sip:0[1-9][0-9]+@.*") { xlog("L_NOTICE", "OpenSER: Ligacao Nacional: $fU -> $tu\n"); } else { ##-- # Internacional ##-- if (uri=~"sip:00[1-9][0-9]+@.*") { xlog("L_NOTICE", "OpenSER: Ligacao Interacional: $fU -> $tu\n"); }; }; route(5); exit; }; if (!lookup("location")) { append_hf("P-hint: usuario nao encontrado\r\n"); sl_send_reply("404", "Usuario nao encontrado"); exit; }; append_hf("P_hint: usrloc applied\r\n"); t_on_failure("1"); route(4); route(1); } route[4] { ##-- # Secao de travessia do nat ##-- if (isflagset(5)) { setbflag(6); }; if (isflagset(5) || isbflagset(6) || isbflagset(7)) { use_media_proxy(); xlog("L_ERR", "Rota[4]: Travessia de NAT mediaproxy\n"); }; } route[5] { ##-- # Encaminhamento para a rede publica (PSTN) ##-- xlog("L_ERR", "Route[5]: Relaying to PSTN\n"); rewritehostport("201.32.70.21"); route(4); route(1); } route[6] { force_rport(); if (nat_uac_test("19")) { if (method=="REGISTER") { fix_nated_register(); } else { fix_nated_contact(); }; setflag(5); }; } failure_route[1] { if (t_check_status("487")) { exit; }; if (isbflagset(6) || isflagset(5)) { end_media_session(); }; onreply_route[1] { ##-- # Bloco de tratamento de respostas ##-- if ((isflagset(5) || isbflagset(6)) && status=~"(180)|(183)|(2[0-9][0-9])") { use_media_proxy(); # search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes'); } if (isbflagset(6)) { fix_nated_contact(); } } Alcindo Schleder Sistema Processa Acessoria e Comunicações Ltda. MSN: a_schleder(a)hotmail.com Skipe: alcindo_schleder Demétrio P. dos Santos, 705 - 95670-000 - Gramado-RS-Brasil Celular+ 55 54 9966 7591 alcindo(a)processa.org Fone+55 54 3286-1738comercial(a)processa.org -----Anexo incorporado----- _______________________________________________ Users mailing list Users(a)lists.openser.org http://lists.openser.org/cgi-bin/mailman/listinfo/users

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[OpenSER-Users] Openser with freeradius not accounting BYE method
by MagoDaViola 05 Apr '08

05 Apr '08

Untitled Document I´m has a problem that my openser sometimes not accounting bye method and this cause a call without pay. See the openser.cfg: # # $Id: ser.cfg,v 1.25.2.1 2005/02/18 14:30:44 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=4 # debug level (cmd line: -dddddddddd) ##-- # # Uncomment these lines to daemonize fork=yes log_stderror=no # (cmd line: -E) ##-- # # Uncomment these lines to enter debugging mode #fork=no #log_stderror=yes ##-- alias=sip.processavoip.net listen=201.32.70.20 #syn_branch=1 memlog=3 sip_warning=1 server_signature=yes reply_to_via=0 mhomed=0 disable_dns_blacklist=yes unix_sock="/tmp/openser.sock" check_via=0 # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 mpath="/usr/local/lib/openser/modules/" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "mysql.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "auth.so" loadmodule "auth_radius.so" loadmodule "uri.so" loadmodule "domain.so" loadmodule "permissions.so" loadmodule "mediaproxy.so" loadmodule "nathelper.so" loadmodule "acc.so" loadmodule "xlog.so" loadmodule "dbtext.so" loadmodule "uri_db.so" loadmodule "mi_fifo.so" loadmodule "group.so" # ----------------- setting module-specific parameters --------------- # -- mi-fifo params -- modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") # -- usrloc params -- modparam("usrloc", "db_mode", 2) modparam("usrloc", "nat_bflag", 6) modparam("usrloc", "domain_column", "fk_domains") modparam("usrloc", "user_column", "fk_accounts") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) ## -- database params -- modparam("acc|usrloc|group|domain|uri_db|permissions", "db_url", "mysql://root:master@localhost/openser") # -- radius params -- modparam("acc|auth_radius", "radius_config", "/etc/radiusclient-ng/radiusclient.conf") # -- radius permissions -- modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") # -- group params -- modparam("group", "table", "accounts_groups") modparam("group", "user_column", "fk_accounts") modparam("group", "domain_column", "fk_domains") modparam("group", "group_column", "pk_accounts_groups") modparam("group", "use_domain", 0) ## -- acc params -- modparam("acc", "radius_missed_flag", 3) modparam("acc", "radius_flag", 2) #modparam("acc", "radius_extra", "via=$hdr(Via[*]); email=$avp(s:email)") modparam("acc", "log_level", 2) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;dst_user=$rU;dst_domain=$rd") modparam("acc", "radius_extra", "User-Name=$fU;Suffix=$fd;Sip-Translated-Request-URI=$ru") modparam("acc", "service_type", 15) modparam("acc", "failed_transaction_flag", 1) modparam("acc", "report_ack", 1) modparam("acc", "report_cancels", 1) modparam("acc", "log_missed_flag", 2) modparam("acc", "log_flag", 1) ## -- acc params without radius -- #modparam("acc", "db_missed_flag", 3) #modparam("acc", "db_flag", 2) #modparam("acc", "db_table_acc", "call_history") #modparam("acc", "db_table_missed_calls", "call_history_missed") # -- uri_db params -- modparam("uri_db", "subscriber_table", "accounts") modparam("uri_db", "subscriber_domain_column", "fk_domains") modparam("uri_db", "subscriber_user_column", "pk_accounts") # -- domain params -- modparam("domain", "domain_table", "domains") modparam("domain", "domain_col", "pk_domains") # -- nathelper, mediaproxy nd registrar params -- modparam("nathelper", "sipping_bflag", 7) modparam("nathelper", "sipping_from", "sip:pinger@openser.org") modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind NAT modparam("registrar|nathelper", "received_avp", "$avp(i:42)") # to configure nat at openser 1.2, 1.3 or 1.4 modparam("mediaproxy", "mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy", "natping_interval", 30) #modparam("mediaproxy", "sip_asymmetrics", "/etc/openser/sip-clients") #modparam("mediaproxy", "rtp_asymmetrics", "/etc/openser/rtp-clients") #modparam("registrar", "received_avp", "$avp(s:rcv)") # to configure nat at openser 1.2, 1.3 or 1.4 # -- tm params -- # set time for which ser will be waiting for a final response; # fr_inv_timer sets value for INVITE transactions, fr_timer # for all others #modparam("tm", "fr_inv_timer", 15 ) #Tempo maximo do ring antes de atender #modparam("tm", "fr_timer", 1) #modparam("tm", "wt_timer", 1) # ------------------------- request routing logic ------------------- # main routing logic route { # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Muitos Hops"); log(1, "486: Muitos Hops\n"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message to big"); log(1, "513: Message too bi \n"); exit; }; ##-- # Eh feita a gravacao de todas as mensagens para se certificar que todas as mensagens # passem pelo proxy o que eh importante particularmente se as entidades estao usando # um protocolo de transporte diferente ##-- if (method!="REGISTER") { record_route(); }; route(6); ##-- # Secao de finalizacao da chamada ##-- if (method=="BYE" || method=="CANCEL") { end_media_session(); }; ##-- # mensagens subsequentes dentro de um dialogo deveriam tomar um caminho pre-determinado # pela gravacao das rotas (record route) ##-- if (loose_route()) { # marca a logica de roteamento no pedido append_hf("P-hint: rr_enforced\r\n"); if (has_totag() && (method=="INVITE" || method=="ACK")) { xlog("L_ERR", "Main Route: Fixing SDP to $fu $rm (loose route)\n"); if (client_nat_test("3") || isflagset(5)) { setbflag(6); use_media_proxy(); }; }; xlog("L_ERR", "Main Route: Loose Route relaying\n"); route(1); exit; }; if (!uri==myself) { # marca a logica de roteamento no pedido append_hf("P-hint: rr_enforced\r\n"); xlog("L_ERR", "Main Route: Relaing mesage to proxy destination \n"); route(4); route(1); }; ##-- # Verifica baseado na tabela de dominios se a parte host da R-URI comtem um dos dominios # quais o proxy eh responsavel ##-- if (!is_uri_host_local()) { if (is_from_local() || allow_trusted()) { route(4); route(1); } else { sl_send_reply("403", "Forbinden..."); } exit; }; ##-- # A rotina abixo eh chamada para marcar o registro com NAT ##-- if (method=="CANCEL") { route(1); exit; } else if (method=="REGISTER") { route(2); exit; } else if (method=="INVITE") { route(3); exit; }; ##-- # Verifica se existem aliases para este endereco e se eles nao pertencem # a teste dominio encaminhamos para frente ##-- lookup("aliases"); if (uri!=myself) { xlog("L_ERR", "Main Route: aliases relayin to other domain proxy\n"); route(4); route(1); exit; }; ##-- # Verifica se encontra o usuario na base de localizacao ##-- if (!lookup("location")) { sl_send_reply("404", "Usuario nao encontrado"); exit; }; route(1); exit; } route[1] { ##-- # Envie isto agora; use "statefull fowarding" pois funciona de forma confiavel de UDP para TCP # Ativa a contabilizacao das chamadas ##-- xlog("L_ERR", "Route[1]: Relaying message\r\n"); setflag(2); ##-- # Ativa a contabilizacao das chamadas perdidas ##-- setflag(3); t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; } route[2] { ##-- # Tratamento padrao para as mensagens de Register ##-- sl_send_reply("100", "Registrando...."); xlog("L_ERR", "Route[2]: Trying Auth: $fu $si\n"); if (isflagset(5) && client_nat_test("7")) { xlog("L_ERR", "Route[2]: Client behind NAT - flag(6)\n"); setbflag(6); fix_nated_register(); force_rport(); # se quiser usar natping options descomente a linha abaixo # setbflag(7); }; if (!radius_www_authorize("")) { www_challenge("", "0"); xlog("L_ERR", "Route[2]: Client Unauthorized: $fu $si\r\n"); sl_send_reply("401", "Unauthorized"); exit; }; if (!check_to()) { xlog("L_ERR", "Route[2]: Client Unauthorized: $fu $si\r\n"); sl_send_reply("401", "Unauthorized"); exit; } else { xlog("L_ERR", "Route[2]: Authenticate: $fu $si\r\n"); }; consume_credentials(); if (!save("location")) { xlog("L_ERR", "Route[2]: Can't save register location\r\n"); sl_reply_error(); }; } route[3] { ##-- # Tratamento padrao para as mensagens de INVITE ##-- xlog("L_ERR", "Route[3]: INVITE: $fu -> $tu $ci\r\n"); # setflag(1); if (!allow_trusted()) { if (!radius_proxy_authorize("")) { proxy_challenge("", "0"); # Realm will be autogenerated exit; } else if (!check_from()) { sl_send_reply("403", "Usue from=ID"); xlog("L_ERR", "Route[3]: Error 403 $fu -> $tu\r\n"); exit; }; } consume_credentials(); ##-- # Verifica se existem aliases para este endereco e se eles nao pertencem # a teste dominio encaminhamos para frente ##-- lookup("aliases"); if (uri!=myself) { xlog("L_ERR", "Main Route: aliases relayin to other domain proxy\n"); route(4); route(1); }; ##-- # VoIP Local ##-- if (uri=~"sip:[1-9][0-9]+@.*") { xlog("L_NOTICE", "OpenSER: Ligacao VoIP local: $fU -> $tu\n"); } else { ##-- # Nacional ##-- if (uri=~"sip:0[1-9][0-9]+@.*") { xlog("L_NOTICE", "OpenSER: Ligacao Nacional: $fU -> $tu\n"); } else { ##-- # Internacional ##-- if (uri=~"sip:00[1-9][0-9]+@.*") { xlog("L_NOTICE", "OpenSER: Ligacao Interacional: $fU -> $tu\n"); }; }; route(5); exit; }; if (!lookup("location")) { append_hf("P-hint: usuario nao encontrado\r\n"); sl_send_reply("404", "Usuario nao encontrado"); exit; }; append_hf("P_hint: usrloc applied\r\n"); t_on_failure("1"); route(4); route(1); } route[4] { ##-- # Secao de travessia do nat ##-- if (isflagset(5)) { setbflag(6); }; if (isflagset(5) || isbflagset(6) || isbflagset(7)) { use_media_proxy(); xlog("L_ERR", "Rota[4]: Travessia de NAT mediaproxy\n"); }; } route[5] { ##-- # Encaminhamento para a rede publica (PSTN) ##-- xlog("L_ERR", "Route[5]: Relaying to PSTN\n"); rewritehostport("201.32.70.21"); route(4); route(1); } route[6] { force_rport(); if (nat_uac_test("19")) { if (method=="REGISTER") { fix_nated_register(); } else { fix_nated_contact(); }; setflag(5); }; } failure_route[1] { if (t_check_status("487")) { exit; }; if (isbflagset(6) || isflagset(5)) { end_media_session(); }; onreply_route[1] { ##-- # Bloco de tratamento de respostas ##-- if ((isflagset(5) || isbflagset(6)) && status=~"(180)|(183)|(2[0-9][0-9])") { use_media_proxy(); # search_append('Contact:.*sip:[^>[:cntrl:]]*', ';nat=yes'); } if (isbflagset(6)) { fix_nated_contact(); } } -------------------------------------------------------------------------- Alcindo Schleder Sistema Processa Acessoria e Comunicações Ltda. a.. MSN: a_schleder(a)hotmail.com b.. Skipe: alcindo_schleder Demétrio P. dos Santos, 705 - 95670-000 - Gramado-RS-Brasil Celular + 55 54 9966 7591 alcindo(a)processa.org Fone +55 54 3286-1738 comercial(a)processa.org

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[OpenSER-Users] How to modify the "Referred-by" field inside a REFER
by Amelie CAMELIERE 04 Apr '08

04 Apr '08

Hi everybody, Is it possible to modify the "Referred-by" field inside a REFER SIP message? In fact I want to modify the name displayed by the phone when I use a REFER. Thanks for your help Amélie

2 1

[OpenSER-Users] Openser with RTPProxy in a different scenario
by morton.olson 04 Apr '08

04 Apr '08

Hi all! I have a problem with Openser an RTPProxy. I am using nathelper module to establish a statefull connection with RTPProxy in between the both clients. I player around with it now for a while but I cant get it to work. All the examples in the internet a for clients behind NAT, etc... Maybe one of you might help me. I am using Openser 1.3.0 with RTPProxy 20040107 on Debian Linux "Etch". My scenario looks like this: UA(public ip) ----- Firewall (public IP; 192.168.0.1/24) ---- Openser+RTPProxy (192.168.0.5/24) ----- UA (192.168.0.23/24) The Firewall forward Port 5060 to the Openser which is listening on 5060 as well. Also Ports from 60000-61000 are forwarded to the Openser aswell. (RTPProxy was compile with these Ports) I have no access to the firewall and it does not manipulate SIP Messages. Is there a chance to get this to work? If so, I willl post my configuration file to see what I did wrong. Thanks a lot. Greetz Martin -- View this message in context: http://www.nabble.com/Openser-with-RTPProxy-in-a-different-scenario-tp16349… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

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[Serusers] Asterisk + OpenSER integration successful
by satish patel 04 Apr '08

04 Apr '08

Dear ser guys.. I have done asterisk with openser integration after spending 2 and 3 days...not time to get nice cold beer... I am user openser front of Asterisk. All sip client register on OpenSER server and call to local extension without Asterisk. caz Asterisk is a media server which are useing only for PSTN calling. When any sip client dialing outside number means mobile number or any outside number Openser forward call to Asterisk and asterisk forward it on PSTN use the E1 or FXO line. ( In my case it is E1 ). OpenSER only user for SIP users or SIP proxy server it can only handle sip request. Asterisk is Media server which can handle IVR , confrance , VoiceMail, and Meetmee or PSTN application. when call come down from PSTN Asterisk forward it on OpenSER as per dialplan configuration. It can handle 1000 of call and you can use it on large production. [sip_client]------------->[Openser]--------->[Asterisk]----[PSTN] My Openser configuration file. #cat ser.cfg debug=6 fork=yes log_stderror=yes listen=<openser_ipaddress> # INSERT YOUR IP ADDRESS HERE port=5060 children=4 check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database #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/textops.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "/usr/local/lib/ser/modules/auth.so" #loadmodule "/usr/local/lib/ser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line #modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # #modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # #modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); break; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication # if (!www_authorize("iptel.org", "subscriber")) { # www_challenge("iptel.org", "0"); # break; # }; save("location"); break; }; # requests for Media server if(method=="INVITE") { if (uri =~ "sip:0[1-9]@*"){ route(3); break; } } lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; } route[3]{ if (uri =~ "sip:0[1-9]@*"){ log(1, "Forwarding to Asterisk\n"); rewritehostport("<asterisk_ip>:5060"); route(1); break; } } #---------END------------# #Asterisk extension.conf [globals] SERADDRESS=203.124.16.109 [outgoing] exten => 5500,1,Playback(hello-world) exten => 5000,1,Answer() exten => 5000,2,Dial(SIP/5000) exten => 5000,3,Hangup() ; XXXX use to forward call on openser extensions. exten => _XXXX,1,Dial(SIP/${EXTEN}@${SERADDRESS},20,r) $ cat ~/satish/url.txt http://www.linuxbug.org _____________________________________________________________________________________________________ --------------------------------- Explore your hobbies and interests. Click here to begin.

1 0

[Serusers] How to set a global variable in ser.cfg and use it in following route() functions?
by lin9k＠yahoo.com 04 Apr '08

04 Apr '08

Hi, Is ser.cfg allow such feature? I'm thinking to use it to hold some variable that can be used for the life of ser.cfg. Thanks in advance, Ling ____________________________________________________________________________________ Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers ____________________________________________________________________________________ You rock. That's why Blockbuster's offering you one month of Blockbuster Total Access, No Cost. http://tc.deals.yahoo.com/tc/blockbuster/text5.com

2 1

[Serusers] calling rewriteuser and rewriteuri from a module?
by lin9k＠yahoo.com 04 Apr '08

04 Apr '08

Hi, I want to provide a module of our own. How do I call rewriteuser and rewriteuri functions from a module? Thanks in advance, Ling ____________________________________________________________________________________ Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers ____________________________________________________________________________________ You rock. That's why Blockbuster's offering you one month of Blockbuster Total Access, No Cost. http://tc.deals.yahoo.com/tc/blockbuster/text5.com

2 1

[OpenSER-Users] 1.3.x package?? forLinux??
by Rey Mulero 04 Apr '08

04 Apr '08

Hello, I am new to OpenSER. I wanted to know if there was a package already compiled to support a P4 Intel machine. I was looking in the website but it really does not spell it out. I see file names openser-1.3.0-tls_linux_i386.tar.gz and openser-1.3.0-notls_linux_i386.tar.gz I am wondering what the difference is? What does tls or notls stand for? Your help is appreciated. Thank you Rey ____________________________________________________________________________________ Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping

2 1

[Serusers] Asterisk + SER call forwading
by satish patel 04 Apr '08

04 Apr '08

Dear all I want to achive forwading call from SER to Asterisk system i have read about it on voip-info real-time integration of SER & asterisk but its base on mysql i dont have mysql database server i just want to use asterisk as PSTN media means all SIP register on SER and "0XXXXXXXXXXX" number goes thorugh asterisk Ser and asterisk runing on defferent machine. I have put this code in my ser.cfg.sample file.. [globals] SERADDRESS=XXX.XXX.XXX.XXX:5060 [context] exten => <yourexten>,1,Dial(SIP/${EXTEN}@${SERADDRESS},20,r) In ser.cfg: if (method == "INVITE") { if (uri =~ "sip:1[0-9]{10}@*"){ log(1, "Forwarding to Asterisk\n"); rewritehostport("XXX.XXX.XXX.XXX:5060"); t_relay(); break; } } But it is not working u got error "User not Found" i dont know way this happending... $ cat ~/satish/url.txt http://www.linuxbug.org _____________________________________________________________________________________________________ --------------------------------- Get the freedom to save as many mails as you wish. Click here to know how.

1 0

[OpenSER-Users] re : Cert advisory and OpenSER V1.3.1
by Jignesh Gandhi 03 Apr '08

03 Apr '08

Hello, I recently upgraded from SER.8.14 to OpenSER 1.3.1 and had a quick question regarding the SIP CERT advisory suite of tests. Has OpenSER been tested against the SIP CERT advisory suite of test cases, that can be downloaded from http://www.ee.oulu.fi/research/ouspg/protos/testing/c07/sip/. This is the SIP CERT website http://www.cert.org/advisories/CA-2003-06.html thanks, Jignesh Gandhi Movius Interactive Corp

2 2

[Serusers] SER for NAT client
by satish patel 03 Apr '08

03 Apr '08

Dear all I have a bit confusen about server and nat its stupid question but i want to clear it wil your suggestions.. I am going to impliment SER for my customers with domain base authentication. Now if my client behind the NAT then how my RTP make connection between two IP phone [SER] / \ / \ / \ [A] [B] In this case RTP between [A] to [B] right But when both [A] and [B] behind the same NAT router then what about RTP what is the path ??? What is the pass of RTP between [A] and [B] in this case [SER] | | | [NAT] / \ / \ [A] [B] $ cat ~/satish/url.txt http://www.linuxbug.org _____________________________________________________________________________________________________ --------------------------------- Now you can chat without downloading messenger. Click here to know how.

2 3

[Serusers] SER and ptime ?
by Mike Trest - Personal 03 Apr '08

03 Apr '08

Hi, I have registering phones (Linksys) that use g711 or g729. I have a media question in a situation when I attempt to do pure-proxy without media gateway or proxy. Because these user are always messing with their adapter settings, I have variable call setup when I forward certain destinaitons because of ptime miss-match between Linksys and destination. Is it possible to accpet / reject / revise the inbound call when the "ptime" is not specifically 20ms or 30ms (or any specific value) before t_relay on to destination? I see the ptime value in SDP but before I embark on journey trying to "manage" call setup based on that, I want to know if others are working with this issue. ..mike..

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[OpenSER-Users] [Fwd: contact uri]
by Amit Vijayvargiya 03 Apr '08

03 Apr '08

2 1

[Serusers] How to return a string value of an exported functions from a module to be used in ser.cfg?
by lin9k＠yahoo.com 03 Apr '08

03 Apr '08

Hi, I am thinking to write a module and export a function to be called from ser.cfg. I need the exported function to return a string value to be used by another function in ser.cfg. How do I do that in ser.cfg? For example: in ser.cfg: function1(my_function()); in my_module: str *my_function(){ ... return my_string; } Thanks in advance, Ling ____________________________________________________________________________________ Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers ____________________________________________________________________________________ You rock. That's why Blockbuster's offering you one month of Blockbuster Total Access, No Cost. http://tc.deals.yahoo.com/tc/blockbuster/text5.com

2 1

[OpenSER-Users] Res: My first OpenSER post
by Flavio Goncalves 02 Apr '08

02 Apr '08

Hi Yakout, Check SerMyAdmin at http://sourceforge.net/projects/sermyadmin. Regards, Flavio E. Goncalves ----- Mensagem original ---- De: Yakout Esmat <yesmat(a)xtra.co.nz> Para: users(a)lists.openser.org Enviadas: Terça-feira, 1 de Abril de 2008 16:08:44 Assunto: [OpenSER-Users] My first OpenSER post Hi All, This is my first port on the OpenSER forum, I come from Asterisk world looking for scalability and redundancy. I have managed to install openSER 1.3 for the first time yesterday on a CentOS 5 test server. I used SVN, which worked fine. My questions is what is the recommendation in regards to web interface. I can see that we have 2 options SerWEB and Admin Web. If there are any links or leads on HOWTO install either one that would be of great help. Which one is recommended for Admin and also for client interface? Does any of them provide FULL admin functionality, as in creating extensions, configuring voice mail etc….? Many thanks for your assistance. No virus found in this outgoing message. Checked by AVG. Version: 7.5.519 / Virus Database: 269.22.1/1352 - Release Date: 3/31/2008 10:13 AM -----Anexo incorporado----- _______________________________________________ Users mailing list Users(a)lists.openser.org http://lists.openser.org/cgi-bin/mailman/listinfo/users

2 1

[Serusers] rtpproxy address filling
by Andres 02 Apr '08

02 Apr '08

Hi, I have a question regarding the way rtpproxy handles 'address filling'. After a session has been created, the rtpproxy pre-fills the caller and callee's addresses and we see that in the rtpproxy output like: pre-filling caller's address with 192.116.246.234:41000 pre-filling callee's address with 192.116.246.234:20000 Then when it sees the actual packets coming in from a different source port, it updates the address and we see it in the log like: callee's address filled in: 192.116.246.234:1024 (RTP) The audio then flows fine both ways. My question is, what would happen it the actual packets came in from a different IP while the rtpproxy was waiting between the state of 'pre-filling' and 'address filled' states? Will the rtpproxy accept such a change that includes a new IP? Will it ignore packets from a different IP and continue the session normally? Or will it abort the session completely? Thanks, Andres http://www.telesip.net

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[Serusers] Fwd: Unix Socket Problem
by frq ser 02 Apr '08

02 Apr '08

Hi Stefan Thanks a lot for your prompt response. I can hear the announcement now. Previously my audio setup was not correct. Best Regards, Frq On Wed, Apr 2, 2008 at 12:11 PM, Stefan Sayer <stefan.sayer(a)iptego.com> wrote: > Hello, > > frq ser wrote: > > > Hi > > > > Thanks for your reply Stefan and my apologies for coming back to this > > issue after such a long time. As you suggested, I included SDP headers in my > > Sipsak INVITE msg and the SEMS seems to like it. I am trying to play the > > announcement using a INVITE message to SEMS at :5070. The SEMS sends a 200 > > OK message, but no audio was being played. In the log, I got these messages > > (just including the important messages)... > > > > .. > > (17358) DEBUG: execute (AmInterfaceHandler.cpp:172): Request OK: > > dispatch it! (17358) DEBUG: getAnnounceFile (Announcement.cpp:79): trying > > '/usr/local/lib/sems/audio/192.168.3.208/200.wav < > > http://192.168.3.208/200.wav>' (17358) DEBUG: getAnnounceFile > > (Announcement.cpp:84): trying '/usr/local/lib/sems/audio/200.wav' > > .. > > .. > > (17358) DEBUG: getCompatiblePayloads (AmSdp.cpp:348): using global > > address: 192.168.3.208 <http://192.168.3.208/> (17358) DEBUG: > > negotiate (AmSession.cpp:262): remote party doesn't support telephone events > > > > Seeing the last line, I then used an Xlite softphone (instead of sipsak) > > to contact SEMS. Again, the call gets established for about 10 seconds and > > then just gets disconnected and no audio can be heard. Nothing in > > > To me the logs look normal. seems like it plays the announcement and then > hangs up, as it should. the default_en.wav (or 200.wav, if you've copied it > there), is only a few seconds long. are you sure that audio is setup > correctly (volume etc), i.e. does the xlite test call work? if you call e.g. > sip:conference@iptel.org <sip%3Aconference(a)iptel.org>, do you hear > something? > > can you run tcpdump -i any or run wireshark to see whether packets flow > between SEMS and xlite? > > you could also PM me a wireshark dump so i could have a look at what's > going on. > > Stefan > >

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Re: [Serusers] Unix Socket Problem
by frq ser 02 Apr '08

02 Apr '08

Hi Thanks for your reply Stefan and my apologies for coming back to this issue after such a long time. As you suggested, I included SDP headers in my Sipsak INVITE msg and the SEMS seems to like it. I am trying to play the announcement using a INVITE message to SEMS at :5070. The SEMS sends a 200 OK message, but no audio was being played. In the log, I got these messages (just including the important messages)... .. (17358) DEBUG: execute (AmInterfaceHandler.cpp:172): Request OK: dispatch it! (17358) DEBUG: getAnnounceFile (Announcement.cpp:79): trying '/usr/local/lib/sems/audio/192.168.3.208/200.wav' (17358) DEBUG: getAnnounceFile (Announcement.cpp:84): trying '/usr/local/lib/sems/audio/200.wav' .. .. (17358) DEBUG: getCompatiblePayloads (AmSdp.cpp:348): using global address: 192.168.3.208 (17358) DEBUG: negotiate (AmSession.cpp:262): remote party doesn't support telephone events Seeing the last line, I then used an Xlite softphone (instead of sipsak) to contact SEMS. Again, the call gets established for about 10 seconds and then just gets disconnected and no audio can be heard. Nothing in the SEMS log (see below) suggests that there is any problem (as far as I can understand). I've checked that the file /usr/local/lib/sems/audio/200.wav is present and I can play it directly in Linux using play command. (The file 200.wav is just a copy of default_en.wav file) I just want to know why the announcement file is not being played and why the call between softphone and SEMS just gets disconnected after a while. I am sure it would be something very basic :( Thanks a lot for your help Frq

2 1

[OpenSER-Users] openser social networking events - April-June 2008
by Daniel-Constantin Mierla 02 Apr '08

02 Apr '08

Hello, in the period of time April - June 2008, there will be a series of openser social networking events organized in different places world wide, during the time some other Open Source or VoIP events take place in the same location. The events are most probably in the form of a dinner, organized together with local representatives of openser community. The events are intended to be rather small size, to allow each one to know the others, in a nice and friendly discussion environment. The primary goal is to strengthen the relations within community and give opportunity to network for collaboration and business. Participation to any of these events is free, everybody pays for its expenses. Anyone interested in openser, sip or voip in general is welcome to join. Next events are: - Bucharest, Romania, April 12, 2008, tied up with ROSDEV2008 (http://www.rosdev.ro - Romanian site only) - Orlando, FL, USA, April 22, 2008, tied up with OPENSER-Asterisk SIP Masterclass (http://www.edvina.net) - Barcelona, Spain, May 6, 2008, tied up with OPENSER-Asterisk SIP Masterclass (http://www.avanzada7.com/) - Vienna, Austria, May 23, 2008, on the way to AsteriskTag (http://www.asterisk-tag.org/wiki/Hauptseite) - Berlin, Germany, May 28, 2008, tied up with LinuxTag (http://www.linuxtag.org) - Amsterdam, Netherlands, June 3, 2008, tied up with VoN.x Europe (http://www.von.com/web/index.php) The hour and place for each event will be announced few days before as we know the number of attending people and we make the reservation. Please send email to me to reserve a place, although you are welcome to come at any time and without prior notice, you may find that there is no more available seat to join the table. If you live in the surroundings of these cities, and want to help with organization, you know a nice and quite place suitable for the event, let me know. Cheers, Daniel -- http://www.asipto.com

1 0

[OpenSER-Users] Building Carrierroute Tables
by Douglas Garstang 02 Apr '08

02 Apr '08

So, I was kind of wondering how those using carrierroute where populating their routes. I was hoping to build a cost table that contained the costs for all our routes and all our carriers, and then construct the carrierroute table as a view derived from the cost table. I guess my SQL (and those around me I asked about it) isn't that good, because I haven't been able to come up with a way to do it yet. Doug. ____________________________________________________________________________________ You rock. That's why Blockbuster's offering you one month of Blockbuster Total Access, No Cost. http://tc.deals.yahoo.com/tc/blockbuster/text5.com

2 1

[OpenSER-Users] My first OpenSER post
by Yakout Esmat 01 Apr '08

01 Apr '08

Hi All, This is my first port on the OpenSER forum, I come from Asterisk world looking for scalability and redundancy. I have managed to install openSER 1.3 for the first time yesterday on a CentOS 5 test server. I used SVN, which worked fine. My questions is what is the recommendation in regards to web interface. I can see that we have 2 options SerWEB and Admin Web. If there are any links or leads on HOWTO install either one that would be of great help. Which one is recommended for Admin and also for client interface? Does any of them provide FULL admin functionality, as in creating extensions, configuring voice mail etc….? Many thanks for your assistance. No virus found in this outgoing message. Checked by AVG. Version: 7.5.519 / Virus Database: 269.22.1/1352 - Release Date: 3/31/2008 10:13 AM

1 0

[OpenSER-Users] Problem activating auth and auth_db modules using openser 1.3.1
by Pascal Maugeri 01 Apr '08

01 Apr '08

Hi My 1.3.1 server does not start when I activate the modules auth and auth_db modules with the following configuration: loadmodule "auth.so" loadmodule "auth_db.so" [...] modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "user_column", "username") modparam("auth_db", "password_column", "password") modparam("auth_db", "domain_column", "domain") modparam("auth_db", "db_url", "mysql://openser:openserrw@192.168.1.4 /openser") modparam("auth_db", "load_credentials", "") There is no error in the log file but no other module is loaded whenever I enable auth and auth_db. Is there anything else I should enable to configure these two modules ? When I disable the loading of these two modules everything works fine. Regards, Pascal

3 3

[Serusers] help required disabling DNS lookup
by geethanjali.csg＠wipro.com 01 Apr '08

01 Apr '08

Hi, I am trying to set up calls between 2 domains without using DNS lookup. UA -> domain1 -> domain2 -> UA I am using static forwarding route the all requests. Requests should be routed using this forward logic only. Both DNS lookup and forward logic are happening. Since I have not configured the DNS, an error "Unresolvable Destination" is seen by the UA. Ser log displays errors related to DNS lookup. I don't want the DNS lookup to happen. Requests should be routed using the forward logic only. Can someone help me solve this problem. Regards, Geethanjali C S Wipro Technologies The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com

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[OpenSER-Users] [openser 1.3] loading module auth.so stop my server
by Pascal Maugeri 01 Apr '08

01 Apr '08

Hi I'm still unable to have a working server 1.3 that loads the module auth.so. It is weird because there is no error message in my log file also I set the debug level to maximum. But after starting the openser service, it says it is successful but there is no running process (pgrep openser does not show anything, when it does if I remove the load auth.so)! How can I get more debugging info about the loading and initialization of auth ? Can anybody share with me his or her configuration file ? I may miss something in mine: loadmodule "sl.so" [...] loadmodule "auth.so" (I believe it is sufficient). Is there any known issue using Redhat Enterprise 4 ? Regards, Pascal

1 0

[OpenSER-Users] Problem in finding invite tsxn for ACK.
by Vikas CG. 01 Apr '08

01 Apr '08

Hi All, We were using the openser server as reg/proxy to test sip gateway. We observed that if we run openser in TCP mode it will add 2 record-routes one for TCP and UDP. But when we put hold UA-client i.e. when we send reinvite the openser server is adding only one record-routes But in ACK for reinvite it is adding 2 record-routes. I am also attaching the sip flow. It would be grateful if any body tells us is this behavior is correct? Because we are depending on record route to find the invite tsxn for ack. Thanks, Vikas C.G.

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Re: [OpenSER-Users] PUA/PUA_MI: More than one Publish sent
by Schumann Sebastian 01 Apr '08

01 Apr '08

Sorry, one short mistake: Of course, after last expire, the expires only removed the a.1207041424.4027.8.0, which was the last one published. The a.1207041424.4027.7.0 from the same publication remains in the table. It has nothing to do with odd and even numbers. Regards Sebastian -----Original Message----- From: Schumann Sebastian Sent: Tuesday, 01. April 2008 10:32 AM To: 'Anca Vamanu' Cc: users(a)lists.openser.org Subject: RE: [OpenSER-Users] PUA/PUA_MI: More than one Publish sent Dear Anca Thank you for the response. The publication of more than one message in your described scenario I understand. In my case, immediately after the first PUBLISH is sent, a second one is sent afterwards. OpenSER does not use the most recent etag for an update, but creates a new one, that is never been removed automatically. If I send an expires=0 message after the publication, only the first published etag is removed. Here my publication script: #!/bin/bash # FIFO command to PUBLISH hard-presence state of user via PUA # #config values OP_SYSUSER="alice(a)10.96.115.129" OP_SIPURI="sip:$OP_SYSUSER" OP_STATE="open" OP_EXPIRES="-1" #build reply fifo file rm -f /tmp/openser_fifo_reply mkfifo /tmp/openser_fifo_reply cat /tmp/openser_fifo_reply& cat >/tmp/openser_fifo <<EOF :pua_publish:openser_fifo_reply $OP_SIPURI $OP_EXPIRES presence application/pidf+xml . <?xml version='1.0' encoding='UTF-8'?><presence xmlns='urn:ietf:params:xml:ns:pidf' xmlns:dm='urn:ietf:params:xml:ns:pidf:data-model' xmlns:rpid='urn:ietf:params:x ml:ns:pidf:rpid' xmlns:c='urn:ietf:params:xml:ns:pidf:cipid' ntity='$OP_SIPURI'><tuple id='gigpoej8'><status><basic>$OP_STATE</basic></status><rpid:user-input> Har d-State</rpid:user-input></tuple><dm:person id='0jfqoiet'><rpid:activities><rpid:unknown/></rpid:activities><dm:note >Hard-State</dm:note></dm:person></presence> EOF And my expires script: #!/bin/bash # FIFO command to PUBLISH hard-presence state of user via PUA # #config values OP_SYSUSER="alice(a)10.96.115.129" OP_SIPURI="sip:$OP_SYSUSER" OP_STATE="closed" OP_EXPIRES="0" #build reply fifo file rm -f /tmp/openser_fifo_reply mkfifo /tmp/openser_fifo_reply cat /tmp/openser_fifo_reply& cat >/tmp/openser_fifo <<EOF :pua_publish:openser_fifo_reply $OP_SIPURI $OP_EXPIRES presence application/pidf+xml . <?xml version='1.0' encoding='UTF-8'?><presence xmlns='urn:ietf:params:xml:ns:pidf' xmlns:dm='urn:ietf:params:xml:ns:pidf:data-model' xmlns:rpid='urn:ietf:params:x ml:ns:pidf:rpid' xmlns:c='urn:ietf:params:xml:ns:pidf:cipid' ntity='$OP_SIPURI'><tuple id='giqpoej8'><status><basic>$OP_STATE</basic></status><rpid:user-input> Har d-State</rpid:user-input></tuple><dm:person id='0jfqoiet'><rpid:activities><rpid:unknown/></rpid:activities><dm:note > ard-State</dm:note></dm:person></presence> EOF I start first publish, then expire, then publish again and so on. The following etags are created normally: e.g. a.1207041424.4027.1.0 a.1207041424.4027.3.0 a.1207041424.4027.5.0 At some point, another one is created together with the "normal" one, e.g. a.1207041424.4027.7.0 and a.1207041424.4027.8.0 The expires only removed the a.1207041424.4027.7.0 (all odd numbers). The published one with the even number stays forever. I don't know what can be the reason. Here my OpenSER configuration file extract from pua module: modparam("pua", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("pua", "db_table", "pua") modparam("pua", "update_period", 10) modparam("pua", "default_expires", 600) Thanks for help. Best regards Sebastian -----Original Message----- From: Anca Vamanu [mailto:anca@voice-system.ro] Sent: Tuesday, 25. March 2008 3:05 PM To: Schumann Sebastian Cc: users(a)lists.openser.org Subject: Re: [OpenSER-Users] PUA/PUA_MI: More than one Publish sent Hello Sebastian, Indeed sometimes more that one Publish messages are sent. I will explain when and why. It is possible for someone to request through the MI command to publish a state that should be active for 5000 seconds (you probably know that -1 in expires filed means infinite time). But the max expires in presence server is lets say 3600 sec. If no other publish request is received before the first one expires, the record is deleted. So, to respect the requested time, the pua module generates itself another publish when the first one is close to expire just to update the validity time. Now, if for a subsequent mi command you do not sent the etag parameter ( set it to '.' value), the most recent etag will be found in pua module, and a correct Publish message will be sent. I have just now tried the scenario you described and it works perfectly for me. So, I want to ask you to say exactly what you are doing because there must be something you have missed out. Is the presentity uri exactly the same for the first and the subsequent mi requests? After how much time have you sent the mi request that fails? Regards, Anca Vamanu Schumann Sebastian wrote: > Dear all > > I am using OpenSER 1.3.1 branch release R3936. I face some bad > behaviour using pua and pua_mi modules. > > I publish presence states via the management interface. Sometimes, it > happens that multiple PUBLISH message are sent by one request. > Unfortunately, there is no "system" or oder when this occurs. It just > happens after some publications. > > In case of an expiration (Expires=0, E-Tag .) it is not that bad, I > receive the log message: ERROR:presence:update_presentity: No E_Tag match. > > In case of publication (Expires=-1, E-Tag .) it is very bad: After > publication, he receives also two similar messages, and threats both > PUBLISH messages as two different ones. Thus, creates two different > presentity entries with different E-Tags. If the state is expired now > via MI (Expires: -1, E-Tag .) only one of the E-Tags is removed. The > second stays in the presentity table and the user still appears online. > > Any ideas what can be the reason and how to solve it? > > Best regards > Sebastian > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)lists.openser.org > http://lists.openser.org/cgi-bin/mailman/listinfo/users >

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Re: [OpenSER-Users] unable to find dialog for BYE
by Jayesh Nambiar 01 Apr '08

01 Apr '08

Hi, This happens only when the call is disconncted from the called party side. When the call initator disconnects there is no such warning in it. Although the dialog ends properly. w/regards, Jayesh ----- Original Message ---- From: Ovidiu Sas <osas(a)voipembedded.com> To: Jayesh Nambiar <voip_freak(a)yahoo.co.in> Cc: users(a)lists.openser.org Sent: Monday, 31 March, 2008 7:21:35 PM Subject: Re: [OpenSER-Users] unable to find dialog for BYE Are you getting this all the time? What kind of matching algorithm are you using? Please post a SIP trace. There are several scenarios that can trigger this warning and it is difficult to debug without providing extra information about your scenario. Regards, Ovidiu Sas On Wed, Mar 26, 2008 at 6:00 AM, Jayesh Nambiar <voip_freak(a)yahoo.co.in> wrote: > > > > Hi All, > I have recently upgraded to openser 1.3.1 and I am trying to explore the > Dialog module. > I want to maintain the dialog for specific set of calls, so I have defined a > route and I have written setflag(4) in that route so that all calls getting > into that route will have the dialogs maintained. Flag 4 is defined for > dialog module. > > Now when the call initiator hangs up; the dialog starts and ends properly > but when the called party hangs up, I get this message in the syslog: > " unable to find dialog for BYE" > > Can someone please guide me on how to interpret this WARNING and what > measures should be taken to avoid the same. > > Thanks in advance > > w/regards, > Jayesh > ________________________________ > Save all your chat conversations. Find them online. > _______________________________________________ > Users mailing list > Users(a)lists.openser.org > http://lists.openser.org/cgi-bin/mailman/listinfo/users > > Unlimited freedom, unlimited storage. Get it now, on http://help.yahoo.com/l/in/yahoo/mail/yahoomail/tools/tools-08.html/

2 1

[OpenSER-Users] writing provider info in sip_trace mysql table.
by Itamar - IspBrasil 01 Apr '08

01 Apr '08

hi. anyone know if is possible to store information in sip_trace mysql table about the company who have completed the call ? example: outbund_company = xxx.xxx.xxx.xxxx -------------------- Itamar Reis Peixoto e-mail/msn: itamar(a)ispbrasil.com.br skype: itamarjp icq: 81053601 +55 11 4063 5033 +55 34 3221 8599

3 2

Re: [OpenSER-Users] dialog matching criteria
by Jayesh Nambiar 01 Apr '08

01 Apr '08

Hi Bogdan, Thanks for your reply. I tried of acheiving the same as follows: Enabled DB support for dialog module. If the called number is 123456, I first query the dialog table to retreive the count on to_uri and compare it with a certain limit value. If the count is equal to or greater than the limit value, just reply as 403, Forbidden. I am about to finish this today and will let you know the results. I just wanted an expert advice on if the logic is ok and efficient. Also there is a column in the dialog table which is caller_route_set and callee_route_set, can someone please tell me from where these values are retreived from as it stays blank while checking the dlg_list. Also is it possible to insert some custom value in this column to implement any further logic. Thanks, w/regards, Jayesh ----- Original Message ---- From: Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> To: Jayesh Nambiar <voip_freak(a)yahoo.co.in> Cc: users(a)lists.openser.org Sent: Monday, 31 March, 2008 5:34:05 PM Subject: Re: [OpenSER-Users] dialog matching criteria Hi Jayesh, Such support is not available, but it is on the roadmap for 1.4. Regards, Bogdan Jayesh Nambiar wrote: > Hi All, > Is there a way in dialog module that I can keep count of active > dialogs based on certain parameters like ruri, to_uri etc. > For example, if i need to do a check of active dialogs on a called > number which is 123456 and take certain actions if the number of > active dialogs to called number 123456 is greater than or equal to > certain integer value. > > Is the above requirement a possibility in openser? > Any help regarding this will be highly appreciated. > > Thanks in advance. > > w/regards, > Jayesh > > ------------------------------------------------------------------------ > Get the freedom to save as many mails as you wish. Click here to know > how. > <http://in.rd.yahoo.com/tagline_mail_5/*http://help.yahoo.com/l/in/yahoo/mai…> > > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)lists.openser.org > http://lists.openser.org/cgi-bin/mailman/listinfo/users > Forgot the famous last words? Access your message archive online at http://in.messenger.yahoo.com/webmessengerpromo.php

1 0

[OpenSER-Users] Effect of use_domain in registrar module
by Chahn Kim 31 Mar '08

31 Mar '08

Hi, all Currently we are running Openser 1.1 registering ATAs just with user name only. This works fine however it generates minor problem with our Python based monitoring script. The script was written under assumption of non blank domain list. resulting only showing those sessions where the user location table has domain field manually inserted. Before we switch to use_domain parameter to 1, We'd like to know any implicit effect of doing it, as well as some questions answered. Q. 1) Does this setting takes "SIP_DOMAIN" env parameter of server itself or the client ATA must send it to Openser in "REGISTER"? Q. 2) Once use_domain in registrar is set to 1, would there be any impacts on existing ATA entries in location table with blank domain? Q. 3) Are there any other parameter that are required to be changed along with use_domain in registrar module? Any comments will be appreciated. John Chahn Kim email: johnk(a)koreanet.us

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[Serusers] Effect of use_domain in registrar module
by Chahn Kim 31 Mar '08

31 Mar '08

Hi, all Currently we are running Openser 1.1 registering ATAs just with user name only. This works fine however it generates minor problem with our Python based monitoring script. The script was written under assumption of non blank domain list. resulting only showing those sessions where the user location table has domain field manually inserted. Before we switch to use_domain parameter to 1, We'd like to know any implicit effect of doing it, as well as some questions answered. Q. 1) Does this setting takes "SIP_DOMAIN" env parameter of server itself or the client ATA must send it to Openser in "REGISTER"? Q. 2) Once use_domain in registrar is set to 1, would there be any impacts on existing ATA entries in location table with blank domain? Q. 3) Are there any other parameter that are required to be changed along with use_domain in registrar module? Any comments will be appreciated. John Chahn Kim email: johnk(a)koreanet.us

1 0

[OpenSER-Users] Res: OpenSER and radiusclient-ng [Unknown Attribute]
by Flavio Goncalves 31 Mar '08

31 Mar '08

Kyle, Check your radius dictionaries and the order of the includes. You are probably not including OpenSER (SIP) attributes. Regards, Flavio E. Goncalves ----- Mensagem original ---- De: Krom <kyle.romulas(a)gmail.com> Para: users(a)lists.openser.org Enviadas: Segunda-feira, 31 de Março de 2008 16:07:00 Assunto: [OpenSER-Users] OpenSER and radiusclient-ng [Unknown Attribute] Parallel posting this to both CDRTool users and OpenSER users, so apologies to members of both. I have installed the latest version of CDRTool and running it with OpenSER 1.1.1 for the time being. When looking at the tcpdump of the records leaving the OpenSER server via radiusclient-ng I am viewing a few 'Unknown Attribute' statements. Ex: Accounting Status Attribute (40), length: 6, Value: Start Service Type Attribute (6), length: 6, Value: #15 Unknown Attribute (102), length: 6, Value: Unknown Attribute (101), length: 6, Value: Username Attribute (1), length: 26, Value: +15552221111(a)x.x.x.x Calling Station Attribute (31), length: 24, Value: sip:+15552221111@x.x.x.x Called Station Attribute (30), length: 32, Value: sip:+15553334444@x.x.x.x Unknown Attribute (107), length: 35, Value: Accounting Session ID Attribute (44), length: 14, Value: 0d487415(a)mysipserver.com Unknown Attribute (104), length: 10, Value: Unknown Attribute (105), length: 7, Value: Unknown Attribute (103), length: 6, Value: Unknown Attribute (214), length: 15, Value: Unknown Attribute (215), length: 6, Value: Unknown Attribute (216), length: 32, Value: Event Timestamp Attribute (55), length: 6, Value: Thu Mar 27 11:01:23 2008 NAS IP Address Attribute (4), length: 6, Value: x.x.x.x NAS Port Attribute (5), length: 6, Value: 5060 Accounting Delay Attribute (41), length: 6, Value: 00 secs I have looked into the dictionary.ser and each of the Attributes are defined and the dictionary file that is defined in radiusclient.conf has the $INCLUDE dictionary.ser statement. Any hints, ideas or links would be appreciated. -- -- Krom kyle.romulas(a)gmail.com -----Anexo incorporado----- _______________________________________________ Users mailing list Users(a)lists.openser.org http://lists.openser.org/cgi-bin/mailman/listinfo/users

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[OpenSER-Users] OpenSER and radiusclient-ng [Unknown Attribute]
by Krom 31 Mar '08

31 Mar '08

Parallel posting this to both CDRTool users and OpenSER users, so apologies to members of both. I have installed the latest version of CDRTool and running it with OpenSER 1.1.1 for the time being. When looking at the tcpdump of the records leaving the OpenSER server via radiusclient-ng I am viewing a few 'Unknown Attribute' statements. Ex: Accounting Status Attribute (40), length: 6, Value: Start Service Type Attribute (6), length: 6, Value: #15 Unknown Attribute (102), length: 6, Value: Unknown Attribute (101), length: 6, Value: Username Attribute (1), length: 26, Value: +15552221111(a)x.x.x.x Calling Station Attribute (31), length: 24, Value: sip:+15552221111@x.x.x.x Called Station Attribute (30), length: 32, Value: sip:+15553334444@x.x.x.x Unknown Attribute (107), length: 35, Value: Accounting Session ID Attribute (44), length: 14, Value: 0d487415(a)mysipserver.com Unknown Attribute (104), length: 10, Value: Unknown Attribute (105), length: 7, Value: Unknown Attribute (103), length: 6, Value: Unknown Attribute (214), length: 15, Value: Unknown Attribute (215), length: 6, Value: Unknown Attribute (216), length: 32, Value: Event Timestamp Attribute (55), length: 6, Value: Thu Mar 27 11:01:23 2008 NAS IP Address Attribute (4), length: 6, Value: x.x.x.x NAS Port Attribute (5), length: 6, Value: 5060 Accounting Delay Attribute (41), length: 6, Value: 00 secs I have looked into the dictionary.ser and each of the Attributes are defined and the dictionary file that is defined in radiusclient.conf has the $INCLUDE dictionary.ser statement. Any hints, ideas or links would be appreciated. -- -- Krom kyle.romulas(a)gmail.com

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[OpenSER-Users] compile Openser 1.3.1 with module carrierroute
by Samael - 31 Mar '08

31 Mar '08

Hi, I tried to compile Openser 1.3.1 with module carrierroute on CentOS 5 in a 64 bits architecture. I compiled the confuse library and when I tried to compile Openser 1.3.1 with carrierroute module show me this error # make prefix=/ include_modules="mysql snmpstats carrierroute" modules ... /usr/bin/ld: /usr/local/lib/libconfuse.a(confuse.o): relocation R_X86_64_32 against `a local symbol' can not be used when making a shared object; recompile with -fPIC /usr/local/lib/libconfuse.a: could not read symbols: Bad value collect2: ld devolviÃ³ el estado de salida 1 make[1]: *** [carrierroute.so] Error 1 make[1]: se sale del directorio `/fuentes/openser/openser-1.3.1-tls /modules/carrierroute' make: *** [modules] Error 2 Can you help me? . If you need more information ask me . Best regards

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[OpenSER-Users] unable to find dialog for BYE
by Jayesh Nambiar 31 Mar '08

31 Mar '08

Hi All, I have recently upgraded to openser 1.3.1 and I am trying to explore the Dialog module. I want to maintain the dialog for specific set of calls, so I have defined a route and I have written setflag(4) in that route so that all calls getting into that route will have the dialogs maintained. Flag 4 is defined for dialog module. Now when the call initiator hangs up; the dialog starts and ends properly but when the called party hangs up, I get this message in the syslog: " unable to find dialog for BYE" Can someone please guide me on how to interpret this WARNING and what measures should be taken to avoid the same. Thanks in advance w/regards, Jayesh Save all your chat conversations. Find them online at http://in.messenger.yahoo.com/webmessengerpromo.php

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[OpenSER-Users] [FEATURE REQ] allow does_uri_exist() to test against other variables besides $ru
by Mike Fedyk 31 Mar '08

31 Mar '08

Hi, Is there a better way to test if a call is from a registered user without authentication than the code below? If not, then let me request that does_uri_exist() take an argument specifying the variable to test against. Mike if (src_ip != 'x.x.x.28' && !lookup("location")) { if (registered("location")) { xlog("L_NOTICE", "cs<$cs> si<$si:$sp> ci<$ci> route3: matched registered()\n"); } if (!registered("location")) { xlog("L_NOTICE", "cs<$cs> si<$si:$sp> ci<$ci> route3: matched !registered()\n"); } ### send DIDs to asterisk directly because it doesn't handle unauthenticated calls via a proxy properly #change request uri to from uri so that does_uri_exist() can test against it $var(orig_ru) = $ru; $ru = $fu; if (does_uri_exist()) { xlog("L_NOTICE", "cs<$cs> si<$si:$sp> ci<$ci> route3: matched does_uri_exist()\n"); } if (!does_uri_exist()) { xlog("L_NOTICE", "cs<$cs> si<$si:$sp> ci<$ci> route3: matched !does_uri_exist()\n"); } #10 digit DID and is not from one of our users if ($var(orig_ru) =~ "^sip:[0-9]{10}" && !does_uri_exist()) { xlog("L_NOTICE", "cs<$cs> si<$si:$sp> ci<$ci> route3: 302 redirect to asterisk\n"); $var(dstredir) = "sip:" + $tU + "@x.x.x.28"; $ru=$var(dstredir); sl_send_reply("302", "Moved Temporarily"); exit; #route(1); }; #END DIDs to asterisk and change request uri back $ru = $var(orig_ru); }

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[OpenSER-Users] dialog matching criteria
by Jayesh Nambiar 31 Mar '08

31 Mar '08

Hi All, Is there a way in dialog module that I can keep count of active dialogs based on certain parameters like ruri, to_uri etc. For example, if i need to do a check of active dialogs on a called number which is 123456 and take certain actions if the number of active dialogs to called number 123456 is greater than or equal to certain integer value. Is the above requirement a possibility in openser? Any help regarding this will be highly appreciated. Thanks in advance. w/regards, Jayesh Chat on a cool, new interface. No download required. Go to http://in.messenger.yahoo.com/webmessengerpromo.php

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[OpenSER-Users] nat_uac_test("31") not enough to detect with some consumer voip passthrough routers
by Mike Fedyk 31 Mar '08

31 Mar '08

Hi, Some of our customers have been connecting behind NAT routers that mangle the sip headers but don't keep the ports open. I've added checks on the source port to catch this case. Has anyone done something similar or better? It'd be nice if this could be added as a flag to nat_uac_test() if it is a common test and there isn't already a better way. Here is the code I've used to handle my registers. I need to do more testing once I can get some of these routers in my lab to see how this affects call processing, not just registering. Mike if (!search("^Contact:\ +\*") && nat_uac_test("7")) { xlog("L_INFO", "M=$rm RURI=$ru F=$fu IP=$si:$sp CS=$cs route2: matched nat_uac_test7\n"); xlog("L_NOTICE", "M=$rm RURI=$ru F=$fu IP=$si:$sp CS=$cs route2: setbflag(6)\n"); setbflag(6); xlog("L_NOTICE", "M=$rm RURI=$ru F=$fu IP=$si:$sp CS=$cs route2: fix_natted_register()\n"); fix_nated_register(); #xlog("L_INFO", "M=$rm RURI=$ru F=$fu IP=$si:$sp CS=$cs route2: force_rport()\n"); #force_rport(); }; #Some NAT devices modify SIP headers to but do not keep the port mappings open, also some UAs detect report the #public IP address. In both cases we need nat ping. Exceptions for default ports. These ports if used as #source ports can be assumed to be at the very least port forwarded. if (!isbflagset(6) && $sp != '5060' && $sp != '5061') { xlog("L_NOTICE", "M=$rm RURI=$ru F=$fu IP=$si:$sp CS=$cs route2: matched !isbflagset(6) && ($sp != '5060' && $sp != '5061')\n"); setbflag(6); };

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[OpenSER-Users] Branch/transaction flags not being inherited by BYE?
by Dan-Cristian Bogos 31 Mar '08

31 Mar '08

Hi All, I have noticed something strange in the version I am running, cannot find the branch/transaction (tried them both) flags in the BYE messages (one app: end_media_session if previously used). Is that a known issue? In the INVITE and 200 OK the flags are there. My version: openser -V version: openser 1.3.1-tls (i386/linux) flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, DISABLE_NAGLE, USE_MCAST, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. svnrevision: 2:3938M @(#) $Id: main.c 3590 2008-01-28 17:46:56Z bogdan_iancu $ main.c compiled on 19:06:15 Mar 19 2008 with gcc 4.1.2 Ta, DanB

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[OpenSER-Users] mailing lists domain migration completed
by Daniel-Constantin Mierla 31 Mar '08

31 Mar '08

Hello, a bit later than previously announced, all mailing lists hosted by openser project work only with domain 'lists.openser.org', the alternative available in the past ('openser.org') has been disabled. You should get a mail delivery error if you use the old domain -- resend to list_name(a)lists.openser.org Cheers, Daniel

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[Serusers] SER performance bottleneck ?
by fang, tian 31 Mar '08

31 Mar '08

Hi dear friends, We tried to benchmarking IPtel SER 0.9.6 on Intel xeon CPU based computer recently,but we met a problem about poort performance.For example on a computer configured with Intel 2*sossaman CPU (four cores)+2G ddr2-400 memory+100M Ethernet card, we didn't configure any database,and registered with only 1-2 users,and we use SIPP 3.0 as the call generator ,but we can only see about 3,000 calls per second for a very simply built-in scenario in SIPP(UAC+UAS),and the CPU idle percentage is about 70-80%.if we tried to increase the call rate ,we got failure calls. We tried to enlarge the memory to 768M for SER,and limited the ser processes number on each port from 8 to 4,and we also made some improvement on config.h & Makefile.defs,but didn't get big improvement on performance . If anybody could help me on this,I will appreciate a lot. Thanks Fang tian.

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Re: [OpenSER-Users] LCR Gateway Priority....
by Douglas Garstang 31 Mar '08

31 Mar '08

>Douglas Garstang writes: > >> Juha, not sure I understand you. It sounds like your suggesting >> exactly the scenario I described in my post, which really seems like >> a less than ideal what to do it. It means a lot of unecessary rows in >> the lcr table. > >if you need to add those rows by hand (i.e., you don't have any >management system) then check if carrierroute module could be a better >match for you. > >-- juha Juha, I spent some time trying to understand and get the carrierroutefunctionality working today, and it looks like it doesn't solve myproblem either. It appears that the default carrier cannot be changed during a call,which is no good. If we assumed that the domain was our carrier, thenit might be ok because we can call lc_rewrite_uri to switch domains,but once we've selected a new carrier, the gateways are selected basedon the 'prob' column in the db. This also isn't what we need. We needmore control than that. The LCR module actually works better. It the gatways had a priority, we'd be happy! I'm surprised no one has written a better module for this. In the realworld, you have multiple POPs, your carriers also have multiple POP's,multiple gateways. In the event you get a failure with one carrier, youprobably want to switch to the next cheapest carrier, not anothergateway within the same carrier (as it will probably just fail again).When selecting a carrier gateway, you need to choose the best POP touse before you select the trunk. Doug. ____________________________________________________________________________________ Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs

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Re: [OpenSER-Users] LCR... Too many entries
by Douglas Garstang 31 Mar '08

31 Mar '08

> >Douglas Garstang writes: > >> Ok, can someone please tell me why 4999 LCR entries is too many? >> What's the limit? Is it hard coded in the OpenSER source? Why? > >there is a constant in lcr_mod.h that tells how many lcr entries cached >lcr table can hold: > >#define MAX_QUERY_SIZE 512 > >you can increase its value, but as i have said, lcr module was not >designed for the kind of inter-operator application that you have. Ok... what about carrierroute? Are there are docs or examples available besides the very limited module specs at openser.org? It seems like if you want to change carriers you have to hard code it in cr_tree_rewrite_uri(). Wouldn't a next_carrier() type function make more sense? Doug. ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ

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Re: [Serusers] SER2.0 xml body size in NOTIFY
by Vaclav Kubart 31 Mar '08

31 Mar '08

Hi Pravin, no, there is no way how to tell SER which information propagate in NOTIFYs and which not. I still think that it is problem on your user agent which doesn't publish/unpublish information correctly. I have seen similar behaviour when a client was ignoring SIP-Etag returned by presence server in PUBLISH response. The same SIP-Etag should be used in all consequnt PUBLISHes from that user agent (see RFC 3903 for details). with regards, Vaclav On Po, bře 31, 2008 at 12:19:48 +0530, pravin.swami(a)wipro.com wrote: > HI... Vaclav , Thanks for reply . > > The problem statement still coming when user changes its state > Online/offline/away . Is there any configurable way in ser presence > > Regards, > pravin > -----Original Message----- > From: Vaclav Kubart [mailto:vaclav.kubart@iptel.org] > Sent: Monday, March 31, 2008 11:55 AM > To: Pravin Sidram swami (WT01 - MCE-Mobile Devices) > Cc: serusers(a)lists.iptel.org > Subject: Re: [Serusers] SER2.0 xml body size in NOTIFY > > Hi, > it seems to me that your PUA doesn't 'unpublish' its presence document when it > goes down. > > SER has to put all known information about presentity into NOTIFY - it can't > decide itself to send some data and some not. > > Vaclav > > On Pá, bře 28, 2008 at 09:36:42 +0530, pravin.swami(a)wipro.com wrote: > > Hi, > > > > > > > > I am facing a problem with growing size of XML body in NOTIFY > > request that carries presence information. > > > > The situation I have is: > > > > 1. PUA1 & PUA2 are two presence user agents subscribed to SER (2.0) > > presence server. > > > > 2. PUA2 comes online. PUA1 gets a NOTIFY. > > > > 3. PUA2 goes down & comes online again. PUA2 gets a NOTIFY with bigger > > XML body than step 2. > > > > 4. Step 3 is repeated many times. The NOTIFY that reaches PUA2 keeps on > > growing in size. > > > > > > > > Is there a way to configure SER server to restrict the size of XML body > > it sends in the NOTIFY request? > > > > Please help us come out of this problem. > > > > > > > > Regards, > > > > -Pravin > > > > > > > > > > The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. > > > > WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. > > > > www.wipro.com > > > > > _______________________________________________ > > Serusers mailing list > > Serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > > > The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. > > WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. > > www.wipro.com

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[Serusers] SER2.0 xml body size in NOTIFY
by pravin.swami＠wipro.com 31 Mar '08

31 Mar '08

Hi, I am facing a problem with growing size of XML body in NOTIFY request that carries presence information. The situation I have is: 1. PUA1 & PUA2 are two presence user agents subscribed to SER (2.0) presence server. 2. PUA2 comes online. PUA1 gets a NOTIFY. 3. PUA2 goes down & comes online again. PUA2 gets a NOTIFY with bigger XML body than step 2. 4. Step 3 is repeated many times. The NOTIFY that reaches PUA2 keeps on growing in size. Is there a way to configure SER server to restrict the size of XML body it sends in the NOTIFY request? Please help us come out of this problem. Regards, -Pravin The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com

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[Serusers] Help with attended (consultative) call transfer problem
by Steven C. Blair 30 Mar '08

30 Mar '08

Hello: We have been having persistent problems with consultative (attended) call transfer and I'd like to ask the list for some input. We have three Cisco 2821 gateways each with two PRIs connecting to Verizon Centrex service. The route advance sequence defined by Verizon results in all calls entering our VoIP environment entering via gateway #1. If no channels exist calls fail over to gateway #2 and finally #3. For load balancing purposes our SER 0.9.7-pre3 proxy will send outbound calls to gateway #3, then #2 and finally #1. We are using Cisco 79x0 phones running SIP load 7.3. All other types of calls including blind transfers work. Call transfers in which both call legs traverse the same gateway work. Only transfers where each call leg traverses different gateways fail. Given the inbound and outbound routing described above it is easy to see how most transfers fail. From a protocol perspective the transfer follows RFCs. I have ethereal traces if anyone needs them. Cisco has worked on this problem and is now suspecting that a transfer will only work if both call legs traverse the same gateway. It is unclear if this is technically correct, a configuration issue in our gateway or a mistake. To further complicate the issue calls from the PSTN, through a gateway to an IP phone which is transfers to another IP phone on the same proxy work. Failures only happen when the second call leg is to the PSTN and traverse a different gateway. Has anyone on this list seen this issue or have any input whatsoever? Thanks,Steve Senior Network Engineer, Information Systems and Computing Networking and Telecommunications , Suite 221A /6228 University of Pennsylvania Voice:215-573-8396 FAX:215-898-9348

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[Serusers] Path header field
by Ladislav Andel 30 Mar '08

30 Mar '08

Hi, does SER support Path header field as described in RFC3327? Thanks, Lada

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[Serusers] Need way to specify SOURCE IP address on packets of SIP messages being emitted by SER
by Frank Durda IV 30 Mar '08

30 Mar '08

I have a problem that I thought SER could solve but now that I need it, the method isn't obvious. First, as an example, let's say I have three independent organizations running Asterisk boxes, which send me the SIP and RTP audio for calls. Let's call them Company A Asterisk box at 31.1.2.3 Company B Asterisk box at 42.4.5.6 Company C Asterisk box at 53.7.8.9 So they are one side of the SER box all communicating SIP to a single interface. On the other side is a TDM PSTN switch capable of accepting SIP calls. The PSTN switch has exactly one IP address, which we will call 101.2.3.4. Inside the PSTN switch, for calls originating in TDM headed to SIP, a "Route List" directs calls for a given trunk group (everything is visualized in TDM terms even if the terms don't really apply) to a specific IP address, where the SIP INVITE should go for this call. In the case of a SIP to TDM call, when an INVITE comes in, the PSTN switch looks through that list of Routes that point to IP addresses and when it finds an IP address match, it knows which trunk group to associate with this call for billing. This also acts as a simple friend/foe mechanism, because if there is no route pointing at your SIP source, it rejects your INVITES. So, if Company A, B and C were communicating directly with the PSTN switch (no SER), you would simply need three trunk groups where each IP address pointed directly to the Asterisk boxes at those companies, like this: TGN SIP IP 1111 -> 31.1.2.3 2222 -> 42.4.5.6 3333 -> 53.7.8.9 and all would be well. However, when you insert a SER box between those the Asterisk boxes belonging to those companies and the PSTN switch, problems arise. I originally decided to address the issue by establishing two groups of three IP addresses, one set that the Asterisk servers would individually send their INVITEs to, and three that the PSTN switch could send its INVITES to. SER already supports the idea of listening on numerous IP addresses, so that was easy (although the documentation for "listen" and "alias" is weak and a code review was needed to determine what alias actually did, something other than what the "alias" option of an ifconfig command does...). Anyway, we have a setup like this: interface interface interface em1 on SER em0 on SER sig0 on PSTN 10.1.0.0/16 10.2.0.0/16 101.2.3.0/28 Company A 31.1.2.3 ---- 10.1.2.3 SER 10.2.0.1 ---- 101.2.3.4 Company B 42.4.5.6 ---- 10.1.5.6 SER 10.2.0.2 ---- 101.2.3.4 Company C 53.7.8.9 ---- 10.1.8.9 SER 10.2.0.3 ---- 101.2.3.4 Inside the ser.cfg, it is simple enough to rewrite the "host address" (where you want the packet to go) to reflect the new destination once that is determined, and for TDM to SIP calls work okay from the perspective of the PSTN switch, but then it really never cared whether the SER box was there or not. As long as the PSTN switch knows who to bill, it is vaguely happy. To accomplish that, the routes inside the PSTN switch were altered to point to the SIP IP addresses that the PSTN switch would have contact with, as in: TGN SIP IP 1111 -> 10.2.0.1 2222 -> 10.2.0.2 3333 -> 10.2.0.3 (Changing the IPs in the PSTN switch turns out to be a non-trivial operation.) The more serious problem arises with a call going SIP to TDM, because when the SER box sends a packet on to the PSTN box at 101.2.3.4, it needs to also re-write the SOURCE IP of the packet to reflect the appropriate IP address on the em0 interface, not just the first IP address that is discovered on the interface. What it does right now is send everything out under the first IP address listed on that interface, usually 10.2.0.1, so calls fail to set up properly as the replies go back to company A even if coming from company B or C. Is there a function in SER that lets you set the right address to use on outgoing packets? As all of the IP addresses that would be used are in fact ones set on that interface, it should be just a question of setting the right one, but I haven't found a function for ser.cfg or loadable module that clearly states that it will do this. So in summary, when Company B sends an INVITE to 10.1.5.6 from their 42.4.5.6 system, I need to have SER send an invite to 101.2.3.4 from 10.2.0.2 and never any other address that happens to be on that same interface. And when the reply to the INVITE comes to 10.2.0.2 from 101.2.3.4, I need SER to forward it on to 42.4.5.6 from 10.1.5.6. Note that while I am showing an example with three companies, there will be dozens, even hundreds of unique IPs involved here, hence the enormous netblocks set aside for each interface on the SER box. I'm certainly open to alternative methods to get SER to alter the headers of the packets it forwards exactly as needed, but keep in mind that I have basically zero control over how the PSTN switch does things, and it has exactly one IP address (three actually, but one does all the SIP work) that you send to and it originates from, yet is keenly interested in where packets it receives came from (for billing purposes) so you can't expect any help from the PSTN switch side. So is there a function callable from ser.cfg to force set the SOURCE IP address on SIP messages on a message by message basis? We are currently using ser-0.9.6. Thanks for your assistance, Frank Durda

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[Serusers] Is there any way to modify ser.cfg without rebooting ser?
by lin9k＠yahoo.com 30 Mar '08

30 Mar '08

Hi, Anybody can illuminate how to modify ser.cfg and make the new modified version effective for ser without rebooting ser? Thanks in advance, Ling ____________________________________________________________________________________ Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping

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[Serusers] Problems with links on iptel.org (Getting Started, module docs etc)
by Greger Viken Teigre 30 Mar '08

30 Mar '08

Due to a domain/registrar problem, the onsip.org domain used in referencing a server with SER Getting Started, module docs, etc has caused broken links on iptel.org the last week or so. The plan is to move everything over to iptel.org servers/domain, but I haven't had time yet. I have now updated the referenced links on iptel.org site to use a temporary domain name, so everything should be ok now. Please raise your voice if there are still onsip.org reference lying around on iptel.org. g-)

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[OpenSER-Users] Issue with Openser Serial Forking
by Aadilkhan Maniyar 29 Mar '08

29 Mar '08

Hi All, I am trying to integrate Openser 1.2.3 and Asterisk 1.4.18. The registrations are handled by Openser and Asterisk acts as a B2BUA. In one of the cases the combo should support Serial Forking. I register an user, ua1 from two different contacts say ua1@contact1 first and ua1@contact2 later. Hence the latest registration of ua1 is ua1@contact2. Now when I call ua1, ua1@contact2 gets two calls, one directly from Openser and second from Asterisk (since I am forwarding the INVITEs from openser to Asterisk). I would like to implement serial forking in Openser. Can someone help me out here.. Regards, Aadil

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[Serusers] acc.so issue
by Santiago Gimenez Ocano 28 Mar '08

28 Mar '08

Hi all, I'm trying to use the acc.so module but for some reason I can't make it load in my ser 0.9.7 (x86_64/linux). This is the stderror output: 0(3184) ERROR: load_module: could not open module </usr/local/lib/ser/modules/acc.so>: /usr/local/lib/ser/modules/acc.so: undefined symbol: parse_orig_uri and this is the list of the modules I'm loading in ser.cfg file: 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/uri.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/uri_radius.so" loadmodule "/usr/local/lib/ser/modules/domain.so" loadmodule "/usr/local/lib/ser/modules/mediaproxy.so" loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/avpops.so" loadmodule "/usr/local/lib/ser/modules/permissions.so" loadmodule "/usr/local/lib/ser/modules/xlog.so" loadmodule "/usr/local/lib/ser/modules/auth_radius.so" loadmodule "/usr/local/lib/ser/modules/acc.so" Does any one have experienced this? Does any one can helpme? Regards, Santiago _________________________________________________________________ Blog your life in 3D with Windows Live Writer. http://www.windowslive.com/overview.html?ocid=TXT_TAGLM_Wave2_wl_writer_022…

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Re: [OpenSER-Users] LCR Routing.... Choosing Best Route
by Douglas Garstang 28 Mar '08

28 Mar '08

Thanks. I'll do all that as soon as I find documentation that gives me some clues. :) ----- Original Message ---- From: Alex Hermann <alex(a)speakup.nl> To: users(a)lists.openser.org Sent: Friday, March 28, 2008 12:36:51 AM Subject: Re: [OpenSER-Users] LCR Routing.... Choosing Best Route On Friday 28 March 2008, Douglas Garstang wrote: > Carrier Prefix Cost > --------------------------- > Carrier1 1650 0.3 > Carrier2 165 0.2 > What do people do in this case? Don't use the LCR module to compare, but do it by other means by selecting the longest matching prefix per carrier and then sort the list on price. Store the list of associated uri's in the gw_uri_avp and use next_gw() to traverse them. Alex. _______________________________________________ Users mailing list Users(a)lists.openser.org http://lists.openser.org/cgi-bin/mailman/listinfo/users ____________________________________________________________________________________ Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping

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[OpenSER-Users] calling external phone number
by Sam Wun 28 Mar '08

28 Mar '08

Hi, I have media proxy, openser, and monit installed in FreeBSD. I have also had the x-lite windows client software working with the openser server. Now, if I want to make phone call from home to external phone number thru the Openser server, what else I have to do to accomplish that? Thanks Sam ____________________________________________________________________________________ Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping

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[OpenSER-Users] [openser install] error while creating the openser DB - presence tables
by Pascal Maugeri 28 Mar '08

28 Mar '08

Hi While creating the openser DB (/usr/local/sbin/openserdbctl create) I get the following error: INFO: creating presence tables into openser ... ERROR 1170 (42000) at line 2: BLOB/TEXT column 'rlsubs_did' used in key specification without a key length ERROR: Failed to create rls-presence tables! FYI I've downloaded the source code from 1.3 branch (rev. 3957) and disabled the following modules in my installation: exclude_modules?= jabber cpl-c postgres osp unixodbc avp_radius auth_radius group_radius uri_radius xmpp presence_mwi pua pua_bla pua_mi pua_usrloc pua_xmpp rls mi_xmlrpc perl snmpstats perlvdb ldap carrierroute h350 xcap_client db_berkeley seas (From the original Makefile I only enabled mysql, presence and presence_xml modules) Should I enable any other module ? It does work when I choose to not install the presence related tables. Thanks in advance for any help or suggestion. Regards, Pascal

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[OpenSER-Users] javax.sip.TransactionUnavailableException: no route! error
by shiebar＠post.tau.ac.il 28 Mar '08

28 Mar '08

Hi, Does anyone have a idea what can cause this error message? I have installed OpenSER with WeSIP and small HelloWorld Servlet. When tring to send INVITE message I get the following WeSIP trace: SipProcessor [ExpressMChannel[6]] - An incoming message is being assigned SipProcessor [SipProcessor[4]] - <<<<<<<<< Request Received <<<<<<<<< INVITE sip:shie_sip2@192.168.2.100 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.102:5061;rport;branch=z9hG4bK23208 To: <sip:shie_sip2@192.168.2.100> From: "Shie" <sip:shie_sip@192.168.2.100>;tag=9844 Call-ID: 1206715243-3208-SHIE(a)10.108.3.122 CSeq: 751 INVITE Max-Forwards: 19 User-Agent: NCH Swift Sound Express Talk 3.06 Contact: <sip:shie_sip@192.168.2.102:5061> Allow: INVITE,CANCEL,ACK,BYE,INFO,OPTIONS,NOTIFY,REFER Supported: replaces Content-Type: application/sdp Content-Length: 352 SeasHashTable [SipProcessor[4]] - Inserting hashIdx:9320 label:781080043 into hashTable SipProcessor [SipProcessor[4]] - Transaction [z9hG4bK23208] is new Transaction?? --> true EngineSipMapper [SipProcessor[4]] - Request from 192.168.2.100 mapped to: ConvergedEngine[Standalone].StandardHttpHost[localhost] ConvergedEngineValve [SipProcessor[4]] - Invoking Host:localhost ConvergedHostValve [SipProcessor[4]] - SipSession not found for requestINVITE, trying match against Rules ConvergedHost [SipProcessor[4]] - Mapped request to:/HelloWorldServlet ConvergedContextValve [SipProcessor[4]] - new session created :1206715243-3208-SHIE@10.108.3.122 StandardProxy [SipProcessor[4]] - sip:shie_sip2@192.168.2.100 Added to proxying targets. StandardProxy [SipProcessor[4]] - doProxy: proxying to sip:shie_sip2@192.168.2.100 StandardProxy [SipProcessor[4]] - ========== Request Sent ============== INVITE sip:shie_sip2@192.168.2.100 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.100:5060,SIP/2.0/UDP 192.168.2.102:5061;rport;branch=z9hG4bK23208 To: <sip:shie_sip2@192.168.2.100> From: "Shie" <sip:shie_sip@192.168.2.100>;tag=9844 Call-ID: 1206715243-3208-SHIE(a)10.108.3.122 CSeq: 751 INVITE Max-Forwards: 18 User-Agent: NCH Swift Sound Express Talk 3.06 Contact: <sip:shie_sip@192.168.2.102:5061> Allow: INVITE,CANCEL,ACK,BYE,INFO,OPTIONS,NOTIFY,REFER Supported: replaces Content-Type: application/sdp Record-Route: <sip:192.168.2.100:5060;lr=on;transport=udp> Content-Length: 352 StandardProxy [SipProcessor[4]] - javax.sip.TransactionUnavailableException: no route! SipConnector [SipProcessor[4]] - recycle: Recycling processor SipProcessor[4] ExpressMessageChannel [ExpressMChannel[6]] - Done processing INVITE sip:shie_sip2@192.168.2.100 SIP/2.0 Thanks in advance, Shie

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[OpenSER-Users] How to check if $rd lies in an IP range
by Josh Mahonin 28 Mar '08

28 Mar '08

Hi folks, In my setup, I've got two disjoint subnets (call then A and B) that cannot communicate directly to each other, but devices on each can both communicate to my OpenSER server and Asterisk box (both on their own subnet, C). There is no NAT involved, so I only want to use rtpproxy when it's the case that device from subnet A attempts to call a device on subnet B, or vice-versa. I would ideally not like to use rtp proxy for communication between A-C and A-B (this will enable RTP media between both subnets, but that solution will not scale very well...) I'm attempting do something like this: if (src_ip == a.b.c.d/24 && dst_ip == w.x.y.z/24) use rtp proxy But unfortunately, on an INVITE, after a lookup, dst_ip is set to the OpenSER server. The pseudovariable $rd is set to the value I'd like to check against, but it complains loudly when I attempt to substitute dst_ip for $rd. Is there any way to use avp_check() or the like to verify that the value in $rd lies in a given subnet? I don't want to match just one IP, but a whole range. I found a similar question on the SER mailing list asked several years ago, with no response. Thanks, Josh

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