- 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

1 0

[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

2 1

[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

3 2

[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

4 3

[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 !

3 7

[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

4 4

[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

1 0

[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

1 0

[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)? -- (\__/) (='.'=) (")_(")

4 4

[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

2 1

[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

10 14

[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

5 5

[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

3 3

[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; };

2 1

[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?

2 2

[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

1 1

[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

2 1

[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.

1 0

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 > > >

3 4

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 > > > > >

3 3

[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

2 3

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 > >

1 0

[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

2 3

[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 ?

1 0

[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. >> > > > > >

1 0

[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]

1 0

[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

2 1

[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

2 1

[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.

2 1

[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..

1 0

[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

4 10

[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 > >

1 0

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

3 3

[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.

2 1

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 >

1 0

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

2 1

[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

2 1

[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

1 0

[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

2 1

[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

3 2

[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); }

2 1

[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

2 1

[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); };

4 3

[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

2 2

[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

3 2

[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.

4 4

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

5 5

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

2 1

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

1 0

[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

3 2

[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

2 2

[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

3 2

[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

3 4

[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

2 1

[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-)

1 0

[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

2 6

[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…

1 0

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

1 0

[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

2 1

[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

1 1

[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

1 0

[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

3 6

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

28 Mar '08

This isn't a question specific to LCR module, but it could be an LCR or Carrierroute module issue. Let's say you have to carriers... Carrier Prefix Cost --------------------------- Carrier1 1650 0.3 Carrier2 165 0.2 When LCR or Carrierroute try to match a dialed number against a routing entry, they will try and match as much of the number as possible, which makes sense. However, when you get routes from multiple carriers, they don't always bill on the same prefix boundaries. In my example above, LCR/Carierroute would match against Carrier1 for anything starting with 1650. HOWEVER, since 165 is also applicable (165 means 1650, 1651 and so on), and in this example cheaper, you should route 165 over Carrier 2. What do people do in this case? Doug ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ

4 3

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

28 Mar '08

Right.... minimising cost is priority #1... and in this particular case the cheapest route is NOT used. ----- Original Message ---- From: Andreas Sikkema <h323(a)ramdyne.nl> To: users(a)lists.openser.org Sent: Thursday, March 27, 2008 5:20:08 PM Subject: Re: [OpenSER-Users] LCR Routing.... Choosing Best Route On Mar 28, 2008, at 12:38 AM, Douglas Garstang wrote: > However, when you get routes from multiple carriers, they don't > always bill on the same prefix boundaries. In my example above, LCR/ > Carierroute would match against Carrier1 for anything starting with > 1650. HOWEVER, since 165 is also applicable (165 means 1650, 1651 > and so on), and in this example cheaper, you should route 165 over > Carrier 2. > > What do people do in this case? It's called LCR for a reason, so I think minimizing cost should be priority 1. Once price has stopped being the differentiator other factors could be taken into account. Maybe price and other costs have to aggregated? -- Andreas Sikkema _______________________________________________ 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

2 1

[OpenSER-Users] UAC_REPLACE_FROM()
by Bai Shi 28 Mar '08

28 Mar '08

Hi all, I would like to share some experience with uac_replace_from(). Please anybody correct me if I made any mistake. I have done tons of test with openser 1.2.3 and 1.3.1 and finally found the trend as follows: In 1.2.3 the Header Field From in CANCEL will be the one received from UAC, no matter what you have done with INVITE, it will remember the From: prior to all the changes. Parameter from_restore_mode doesn't apply to this. So you have to write something like if(method=="CANCEL") uac_replace_from("sometext"). However, in 1.3.1. If you set modparam("uac", "from_restore_mode", "auto"). TM is smart enough to change From: in CANCEL for you to match the one in INVITE, even modified by Openser. But the problem in 1.3.1 is if you set from_restore_mode to none, no matter how you write uac_replace_from() to your CANCEL request, nothing will happen. Actually this means in Openser 1.3.1, no matter what you do with uac_replace_from() to CANCEL, nothing will happen, depends on the module param it will be either original message or modified message in INVITE. But you can hardly do anything to CANCEL manually. Bogdan, this is the behavior you intend to? Best Regards, Bai Shi

2 2

Re: [OpenSER-Users] Carrierroute... Do I almost have it???
by Douglas Garstang 28 Mar '08

28 Mar '08

Ok... I can do it if I use cr_user_rewrite_uri("sip:.*@192.168.1.240", "1") in openser.cfg and then change the username from blank to '.*' (WHERE is that documented?? I only worked it out by tracing port 3306!!!). But... this isn't feesible. This means that I have to HARD CODE the name of the user in the openser.cfg... Doug. ----- Original Message ---- From: Douglas Garstang <dougmig33(a)yahoo.com> To: Ovidiu Sas <osas(a)voipembedded.com> Cc: users(a)lists.openser.org Sent: Wednesday, March 26, 2008 9:41:23 AM Subject: Re: [OpenSER-Users] Carrierroute... Do I almost have it??? Ovidiu, Ok, so now it's routing but it's not selecting the carrier by matching the source IP address in the subscriber table against the IP address of the incoming INVITE. Is that even possible? If it's not possible, how do I have a different routing table for each OpenSER system? I was going to use my carrier as my OpenSER instance.... that would have worked... I have these tables... mysql> select * from carrierroute; +----+---------+-------------+--------+------+-------+--------------+----------------+----------------+---------+ | id | carrier | scan_prefix | domain | prob | strip | rewrite_host | rewrite_prefix | rewrite_suffix | comment | +----+---------+-------------+--------+------+-------+--------------+----------------+----------------+---------+ | 1 | 0 | | 1 | 1 | 0 | 200.1.1.1 | | | NULL | | 4 | 0 | | 2 | 1 | 0 | 200.1.1.2 | | | NULL | | 5 | 0 | | 3 | 1 | 0 | 200.1.1.3 | | | NULL | | 9 | 1 | | 1 | 1 | 0 | 100.1.1.1 | | | NULL | | 10 | 1 | | 2 | 1 | 0 | 100.1.1.2 | | | NULL | | 11 | 1 | | 3 | 1 | 0 | 100.1.1.3 | | | NULL | +----+---------+-------------+--------+------+-------+--------------+----------------+----------------+---------+ 6 rows in set (0.00 sec) mysql> select * from route_tree; +----+-----------+ | id | carrier | +----+-----------+ | 1 | Hong Kong | | 0 | default | | 2 | San Jose | +----+-----------+ mysql> select username, domain, cr_preferred_carrier from subscriber; +----------+----------------+----------------------+ | username | domain | cr_preferred_carrier | +----------+----------------+----------------------+ | | 192.168.255.53 | 2 | | | 192.168.1.240 | 1 | +----------+----------------+----------------------+ ----- Original Message ---- From: Ovidiu Sas <osas(a)voipembedded.com> To: Douglas Garstang <dougmig33(a)yahoo.com> Cc: Henning Westerholt <henning.westerholt(a)1und1.de>; users(a)lists.openser.org Sent: Wednesday, March 26, 2008 6:56:15 AM Subject: Re: [OpenSER-Users] Carrierroute... Do I almost have it??? On Wed, Mar 26, 2008 at 2:31 AM, Douglas Garstang <dougmig33(a)yahoo.com> wrote: > > >I am using this module in production with more then 400k routes loaded > >for 8 different providers and it is working like a charm. > >As Henning mentioned, the interface is a little bit cumbersome but usable. > >You can put all the carriers under the same table and prioritize the > >gateway based on domains. Like tis you don't need to hardcode > >anything in the config file. > > > >Carrierroute was designed to deal with a big amount of routing rules > >and it is doing an amazing job. > >LCR was designed to deal with a limited number of routes/gateways but > >it has a different level of flexibility and a more mature interface. > > > >Carrierroute is a brand new module and for sure it will improve in > >flexibility as times go by (just like the lcr did along releases). > > Ovidiu (or anyone else....) > > I think I may almost have it..... If I treat all routes as being in the same > carrier, I can just use cr_rewrite_uri("1","call_id"), and then > cr_rewrite_uri("2","call_id") in failure_route[1] and > cr_rewrite_uri("2","call_id") in failure_route[2], correct? I tested this > and it seems to work. I guess it's up to me then to populate the > rewrite_host column with a gateway belonging to the cheapest carrier for > domain 1, the a gateway from the next cheapest carrier for domain 2 and so > on. Still sound right? finally you got it ;) > Here's where it gets a bit tricky. We have OpenSER running in 6 different > locations. The gateways we use will differ for each location (we always want > to use the closest gateway belonging to a carrier). So, I figured I could > use the 'user' functionality in carrierroute to do this. I could simply put > the domain (IP address) in the domain column, leave the user blank (we don't > have users... calls are forwarded from asterisk), and then associate a > carrier in the cr_preferred_carrier column. You can see all that in my > subscriber table below. > > Doesn't seem to work though. When routing a call, OpenSER complains that it > can't find a default carrier, and then because it seems not to match the > incoming call against an entry in the subscriber table, it can't find a > carrier to use, and fails the call. add a 'default' carrier into the route_tree table > So... how would I be able to use the source IP address of the asterisk box > that sends the calls to OpenSER in the subscriber table so that I could have > different routing entries for each OpenSER system??? > > Here's my db tables... > > > mysql> select * from route_tree; > +----+-----------+ > | id | carrier | > +----+-----------+ > | 1 | Hong Kong | > | 0 | San Jose | > +----+-----------+ > > mysql> select * from carrierroute order by carrier, scan_prefix, domain; > +----+---------+-------------+--------+------+-------+--------------+----------------+----------------+---------+ > | id | carrier | scan_prefix | domain | prob | strip | rewrite_host | > rewrite_prefix | rewrite_suffix | comment | > +----+---------+-------------+--------+------+-------+--------------+----------------+----------------+---------+ > | 1 | 0 | | 1 | 1 | 0 | 200.1.1.1 | > | | NULL | > | 4 | 0 | | 2 | 1 | 0 | 200.1.1.2 | > | | NULL | > | 5 | 0 | | 3 | 1 | 0 | 200.1.1.3 | > | | NULL | > | 9 | 1 | | 1 | 1 | 0 | 100.1.1.1 | > | | NULL | > | 10 | 1 | | 2 | 1 | 0 | 100.1.1.2 | > | | NULL | > | 11 | 1 | | 3 | 1 | 0 | 100.1.1.3 | > | | NULL | > +----+---------+-------------+--------+------+-------+--------------+----------------+----------------+---------+ > > mysql> select id, username, domain, cr_preferred_carrier from subscriber; > +----+----------+----------------+----------------------+ > | id | username | domain | cr_preferred_carrier | > +----+----------+----------------+----------------------+ > | 1 | | 192.168.255.53 | 0 | > | 2 | | 192.168.255.54 | 1 | > +----+----------+----------------+----------------------+ > > > > > > > ________________________________ > Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it > now. Never miss a thing. Make Yahoo your homepage. ____________________________________________________________________________________ Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping

2 1

[OpenSER-Users] OpenSER and radiusclient-ng [Unknown Attribute]
by Krom 27 Mar '08

27 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

1 0

[OpenSER-Users] Rewriting the to header
by shiebar＠post.tau.ac.il 27 Mar '08

27 Mar '08

Hi, Who is inchage of replacing the to header in the message (openser/wesip)? When wesip get the message the to uri is still with the openser IP. Do I need to do something when the request is INVITE request or these openser replace it automatically before he send it to wesip? Please advice. I realy need it soon. Thanks in advance, Shie

3 9

[OpenSER-Users] URI Length
by VoIP Forums www.Go4Calls.com 27 Mar '08

27 Mar '08

Hi All, How i can get the length of called number. i am trying to making routing logic base on length. Please help us how i can get the length i tried the fillowing but did not work. if (uri:len<=7) { rewritehostport("192.168.1.1:5060"); route(1); exit; }; Regards, www.Go4Calls.Com VoIP Forums _________________________________________________________________ Connect to the next generation of MSN Messenger http://imagine-msn.com/messenger/launch80/default.aspx?locale=en-us&source=…

2 1

[Serusers] 转发： SER performance bottleneck ?
by fang, tian 27 Mar '08

27 Mar '08

-------原始邮件------- 发件人： fang, tian 日期： 2008-3-27 12:59:27 收件人： serusers(a)iptel.org 主题： SER performance bottleneck ? 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.

1 0

[OpenSER-Users] tm module fails if RURI is not a SIP URI
by Klaus Darilion 26 Mar '08

26 Mar '08

Hi! I just tried if openser can handle a service URN in the request URI (e.g. urn:service:sos which is used for emergency calls (refer to IETF ECRIT WG)). Do it stateless with forward() works fine. Doing it statefull with tm module and t_relay() fails: DBG:tm:t_newtran: transaction on entrance=(nil) DBG:core:parse_headers: flags=ffffffffffffffff DBG:core:parse_headers: flags=78 DBG:tm:t_lookup_request: start searching: hash=20823, isACK=0 DBG:tm:matching_3261: RFC3261 transaction matching failed DBG:tm:t_lookup_request: no transaction found ERROR:core:parse_uri: bad uri, state 0 parsed: <urn:> (4) / <urn:service:sos> (15) ERROR:core:parse_sip_msg_uri: bad uri <urn:service:sos> DBG:core:set_err_info: ec: 1, el: 3, ei: 'error parsing r-uri' ERROR:tm:new_t: uri invalid ERROR:tm:t_newtran: new_t failed I wonder, is it really necessary for the tm module to understand the syntax of the RURI? Of course it is necessary if routing is done based on the RURI. But if DURI is set, or t_relay("udp:1.2.3.4:5060") is used, then I think it should work with any uri scheme in RURI. regards klaus

2 2

[Serusers] ser with asterisk
by aka 26 Mar '08

26 Mar '08

i want to integrate ser with asterisk.i try to do that in one box.asteriskwill communicate 5060 port and ser will communicate 6070 port.two servers work fine in seperatley.but i want to integrate the asterisk and ser in same machine.please give me some help for this iam very beginner for this. thanx akalanka

2 1

[OpenSER-Users] automatically generated aliases with usrloc and ldap
by Lukasz Stelmach 26 Mar '08

26 Mar '08

Greetings All. I have managed to set up OpenSER with LDAP authentication. Now I would like it to create memory cached aliases each time a user registers. I do not wan to use any persistent storage because I have it already it is my LDAP directory. However, I would like OpenSER to cache those aliases as doing ldap lookup for every message does not look too apealing to me. Let me explain how I see it: if (is_method("REGISTER")) { if(is_present_hf("Proxy-Authorization")) { if (!ldap_search("ldap://sipaccounts/ou=dept,o=TELMARK,c=PL?uid,mail,uidNumber,userPassword?one?(uid=$f)) { #error handling } ldap_result("uid/$avp(s:username)"); ldap_result("userPassword/$avp(s:password)"); ldap_result("mail/$avp(s:longalias)"); ldap_result("uidNumber/$avp(s:numalias)"); if(pv_proxy_authorize("telmark")) { proxy_challenge("telmark", "1"); exit; } } else { proxy_challenge("telmark", "1"); exit; } if(!save("location")) sl_reply_error(); # And here the big thing! } What to do to make save("aliases"); (called twice) save uidNumber and mail (without @domain, these are transformations, I know) as aliases in usrloc? PS. Please cc answers I am not a subscriber. -- Best regards, >Łukasz<

2 1

Re: [OpenSER-Users] Carrierroute... Do I almost have it???
by Douglas Garstang 26 Mar '08

26 Mar '08

Ovidiu, Ok, so now it's routing but it's not selecting the carrier by matching the source IP address in the subscriber table against the IP address of the incoming INVITE. Is that even possible? If it's not possible, how do I have a different routing table for each OpenSER system? I was going to use my carrier as my OpenSER instance.... that would have worked... I have these tables... mysql> select * from carrierroute; +----+---------+-------------+--------+------+-------+--------------+----------------+----------------+---------+ | id | carrier | scan_prefix | domain | prob | strip | rewrite_host | rewrite_prefix | rewrite_suffix | comment | +----+---------+-------------+--------+------+-------+--------------+----------------+----------------+---------+ | 1 | 0 | | 1 | 1 | 0 | 200.1.1.1 | | | NULL | | 4 | 0 | | 2 | 1 | 0 | 200.1.1.2 | | | NULL | | 5 | 0 | | 3 | 1 | 0 | 200.1.1.3 | | | NULL | | 9 | 1 | | 1 | 1 | 0 | 100.1.1.1 | | | NULL | | 10 | 1 | | 2 | 1 | 0 | 100.1.1.2 | | | NULL | | 11 | 1 | | 3 | 1 | 0 | 100.1.1.3 | | | NULL | +----+---------+-------------+--------+------+-------+--------------+----------------+----------------+---------+ 6 rows in set (0.00 sec) mysql> select * from route_tree; +----+-----------+ | id | carrier | +----+-----------+ | 1 | Hong Kong | | 0 | default | | 2 | San Jose | +----+-----------+ mysql> select username, domain, cr_preferred_carrier from subscriber; +----------+----------------+----------------------+ | username | domain | cr_preferred_carrier | +----------+----------------+----------------------+ | | 192.168.255.53 | 2 | | | 192.168.1.240 | 1 | +----------+----------------+----------------------+ ----- Original Message ---- From: Ovidiu Sas <osas(a)voipembedded.com> To: Douglas Garstang <dougmig33(a)yahoo.com> Cc: Henning Westerholt <henning.westerholt(a)1und1.de>; users(a)lists.openser.org Sent: Wednesday, March 26, 2008 6:56:15 AM Subject: Re: [OpenSER-Users] Carrierroute... Do I almost have it??? On Wed, Mar 26, 2008 at 2:31 AM, Douglas Garstang <dougmig33(a)yahoo.com> wrote: > > >I am using this module in production with more then 400k routes loaded > >for 8 different providers and it is working like a charm. > >As Henning mentioned, the interface is a little bit cumbersome but usable. > >You can put all the carriers under the same table and prioritize the > >gateway based on domains. Like tis you don't need to hardcode > >anything in the config file. > > > >Carrierroute was designed to deal with a big amount of routing rules > >and it is doing an amazing job. > >LCR was designed to deal with a limited number of routes/gateways but > >it has a different level of flexibility and a more mature interface. > > > >Carrierroute is a brand new module and for sure it will improve in > >flexibility as times go by (just like the lcr did along releases). > > Ovidiu (or anyone else....) > > I think I may almost have it..... If I treat all routes as being in the same > carrier, I can just use cr_rewrite_uri("1","call_id"), and then > cr_rewrite_uri("2","call_id") in failure_route[1] and > cr_rewrite_uri("2","call_id") in failure_route[2], correct? I tested this > and it seems to work. I guess it's up to me then to populate the > rewrite_host column with a gateway belonging to the cheapest carrier for > domain 1, the a gateway from the next cheapest carrier for domain 2 and so > on. Still sound right? finally you got it ;) > Here's where it gets a bit tricky. We have OpenSER running in 6 different > locations. The gateways we use will differ for each location (we always want > to use the closest gateway belonging to a carrier). So, I figured I could > use the 'user' functionality in carrierroute to do this. I could simply put > the domain (IP address) in the domain column, leave the user blank (we don't > have users... calls are forwarded from asterisk), and then associate a > carrier in the cr_preferred_carrier column. You can see all that in my > subscriber table below. > > Doesn't seem to work though. When routing a call, OpenSER complains that it > can't find a default carrier, and then because it seems not to match the > incoming call against an entry in the subscriber table, it can't find a > carrier to use, and fails the call. add a 'default' carrier into the route_tree table > So... how would I be able to use the source IP address of the asterisk box > that sends the calls to OpenSER in the subscriber table so that I could have > different routing entries for each OpenSER system??? > > Here's my db tables... > > > mysql> select * from route_tree; > +----+-----------+ > | id | carrier | > +----+-----------+ > | 1 | Hong Kong | > | 0 | San Jose | > +----+-----------+ > > mysql> select * from carrierroute order by carrier, scan_prefix, domain; > +----+---------+-------------+--------+------+-------+--------------+----------------+----------------+---------+ > | id | carrier | scan_prefix | domain | prob | strip | rewrite_host | > rewrite_prefix | rewrite_suffix | comment | > +----+---------+-------------+--------+------+-------+--------------+----------------+----------------+---------+ > | 1 | 0 | | 1 | 1 | 0 | 200.1.1.1 | > | | NULL | > | 4 | 0 | | 2 | 1 | 0 | 200.1.1.2 | > | | NULL | > | 5 | 0 | | 3 | 1 | 0 | 200.1.1.3 | > | | NULL | > | 9 | 1 | | 1 | 1 | 0 | 100.1.1.1 | > | | NULL | > | 10 | 1 | | 2 | 1 | 0 | 100.1.1.2 | > | | NULL | > | 11 | 1 | | 3 | 1 | 0 | 100.1.1.3 | > | | NULL | > +----+---------+-------------+--------+------+-------+--------------+----------------+----------------+---------+ > > mysql> select id, username, domain, cr_preferred_carrier from subscriber; > +----+----------+----------------+----------------------+ > | id | username | domain | cr_preferred_carrier | > +----+----------+----------------+----------------------+ > | 1 | | 192.168.255.53 | 0 | > | 2 | | 192.168.255.54 | 1 | > +----+----------+----------------+----------------------+ > > > > > > > ________________________________ > Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it > now. ____________________________________________________________________________________ Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping

1 0

[Serusers] [SPAM] Re: pls. help me on my installation
by Martin Hoffmann 26 Mar '08

26 Mar '08

Spam detection software, running on the system "h1063274.serverkompetenz.net", has identified this incoming email as possible spam. The original message has been attached to this so you can view it (if it isn't spam) or label similar future email. If you have any questions, see the administrator of that system for details. Content preview: ram wrote: > On Mon, Mar 24, 2008 at 3:32 PM, Allan Jardinel <ajardinel2002(a)yahoo.com> > wrote: > > > I'm using ubuntu 7.10 and install ser 0.9.7 and rtp, after the > > installation i test and it can call both users, when i try to restart the > > computer xten can not register anymore so i check the settings and its ok. > > but when i check the /var/run/ser.pid there is no file, also the > > /tmp/ser_fifo has no file so restart the ser using the /etc/ser restart or > > force-reload but still pid and fifo has no file. pls. help me to solve this > > problem.... > > > > > Hi > > you try run manuall with > > ser -c -f config file > > see you see any errrors [...] Content analysis details: (2.3 points, 2.0 required) pts rule name description ---- ---------------------- -------------------------------------------------- 0.3 MAILTO_TO_SPAM_ADDR URI: Includes a link to a likely spammer email 2.0 RCVD_IN_SORBS_DUL RBL: SORBS: sent directly from dynamic IP address [80.218.12.224 listed in dnsbl.sorbs.net]

1 0

Re: [OpenSER-Users] To-tag of 200 for CANCEL
by Andreas Granig 26 Mar '08

26 Mar '08

Corrected list address. Andreas Granig wrote: > Hi Bogdan, > > Yes, and I also read some older discussions about this topic like > http://www.ietf.org/mail-archive/web/sipping/current/msg03071.html but > still I'm quite uneasy regarding the phrasing in RFC3261 saying "that > the To tag of the response to the CANCEL and the To tag in the response > to the original request SHOULD be the same". It doesn't clearly state to > which response it refers - the final 487 for the INVITE or maybe to > possible 180/183 (although I understand the consequences and > implications of parallel forking etc.). > > I currently don't have a trace available, but is the To-tag in the 200 > to the CANCEL really the same as the To-tag in 487 to the INVITE, as the > RFC suggests, or is the 487 just passed through? > > Andreas > > Bogdan-Andrei Iancu wrote: >> Hi Andreas, >> >> Have you went through this: >> >> http://sourceforge.net/tracker/index.php?func=detail&aid=1818469&group_id=1… >> >> There is also a reference to a discussion (on the topic) on sip >> implementers list. >> >> Regards, >> Bogdan >> >> Andreas Granig wrote: >>> Hi, >>> >>> Actually, someone pointed out a previous discussion about that issue. >>> Interesting read. >>> http://www.openser.org/pipermail/users/2007-October/014049.html >>> >>> Andreas >>> >>> Andreas Granig wrote: >>>> Hi, >>>> >>>> One more question regarding the CANCEL processing: >>>> >>>> RFC says in §9.2 regarding the 200 for a CANCEL: >>>> >>>> #+ >>>> This response is >>>> constructed following the procedures described in Section 8.2.6 >>>> noting that the To tag of the response to the CANCEL and the To tag >>>> in the response to the original request SHOULD be the same. >>>> #- >>>> >>>> OpenSER (1.3.1) on the other hand does generate a different To-tag >>>> and doesn't use the one from the response of the original request. >>>> Why is that? Anything we can do about this? >>>> >>>> Best regards, >>>> Andreas >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users(a)lists.openser.org >>>> http://lists.openser.org/cgi-bin/mailman/listinfo/users >>> >> >

1 0

Re: [OpenSER-Users] Carrierroute... Do I almost have it???
by Douglas Garstang 26 Mar '08

26 Mar '08

>I am using this module in production with more then 400k routes loaded >for 8 different providers and it is working like a charm. >As Henning mentioned, the interface is a little bit cumbersome but usable. >You can put all the carriers under the same table and prioritize the >gateway based on domains. Like tis you don't need to hardcode >anything in the config file. > >Carrierroute was designed to deal with a big amount of routing rules >and it is doing an amazing job. >LCR was designed to deal with a limited number of routes/gateways but >it has a different level of flexibility and a more mature interface. > >Carrierroute is a brand new module and for sure it will improve in >flexibility as times go by (just like the lcr did along releases). Ovidiu (or anyone else....) I think I may almost have it..... If I treat all routes as being in the same carrier, I can just use cr_rewrite_uri("1","call_id"), and then cr_rewrite_uri("2","call_id") in failure_route[1] and cr_rewrite_uri("2","call_id") in failure_route[2], correct? I tested this and it seems to work. I guess it's up to me then to populate the rewrite_host column with a gateway belonging to the cheapest carrier for domain 1, the a gateway from the next cheapest carrier for domain 2 and so on. Still sound right? Here's where it gets a bit tricky. We have OpenSER running in 6 different locations. The gateways we use will differ for each location (we always want to use the closest gateway belonging to a carrier). So, I figured I could use the 'user' functionality in carrierroute to do this. I could simply put the domain (IP address) in the domain column, leave the user blank (we don't have users... calls are forwarded from asterisk), and then associate a carrier in the cr_preferred_carrier column. You can see all that in my subscriber table below. Doesn't seem to work though. When routing a call, OpenSER complains that it can't find a default carrier, and then because it seems not to match the incoming call against an entry in the subscriber table, it can't find a carrier to use, and fails the call. So... how would I be able to use the source IP address of the asterisk box that sends the calls to OpenSER in the subscriber table so that I could have different routing entries for each OpenSER system??? Here's my db tables... mysql> select * from route_tree; +----+-----------+ | id | carrier | +----+-----------+ | 1 | Hong Kong | | 0 | San Jose | +----+-----------+ mysql> select * from carrierroute order by carrier, scan_prefix, domain; +----+---------+-------------+--------+------+-------+--------------+----------------+----------------+---------+ | id | carrier | scan_prefix | domain | prob | strip | rewrite_host | rewrite_prefix | rewrite_suffix | comment | +----+---------+-------------+--------+------+-------+--------------+----------------+----------------+---------+ | 1 | 0 | | 1 | 1 | 0 | 200.1.1.1 | | | NULL | | 4 | 0 | | 2 | 1 | 0 | 200.1.1.2 | | | NULL | | 5 | 0 | | 3 | 1 | 0 | 200.1.1.3 | | | NULL | | 9 | 1 | | 1 | 1 | 0 | 100.1.1.1 | | | NULL | | 10 | 1 | | 2 | 1 | 0 | 100.1.1.2 | | | NULL | | 11 | 1 | | 3 | 1 | 0 | 100.1.1.3 | | | NULL | +----+---------+-------------+--------+------+-------+--------------+----------------+----------------+---------+ mysql> select id, username, domain, cr_preferred_carrier from subscriber; +----+----------+----------------+----------------------+ | id | username | domain | cr_preferred_carrier | +----+----------+----------------+----------------------+ | 1 | | 192.168.255.53 | 0 | | 2 | | 192.168.255.54 | 1 | +----+----------+----------------+----------------------+ ____________________________________________________________________________________ Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs

2 1

[OpenSER-Users] To-tag of 200 for CANCEL
by Andreas Granig 26 Mar '08

26 Mar '08

Hi, One more question regarding the CANCEL processing: RFC says in §9.2 regarding the 200 for a CANCEL: #+ This response is constructed following the procedures described in Section 8.2.6 noting that the To tag of the response to the CANCEL and the To tag in the response to the original request SHOULD be the same. #- OpenSER (1.3.1) on the other hand does generate a different To-tag and doesn't use the one from the response of the original request. Why is that? Anything we can do about this? Best regards, Andreas

2 2

[OpenSER-Users] UA Cant Connect To TLS
by Ali Jawad 26 Mar '08

26 Mar '08

Hi All My UA cant connect to tls on my openser server, as per logs everything seems fine ..I don't know why it disconnects in the end ..the log is http://java.pastebin.ca/948774 My TLS settings are /* uncomment the following lines to enable TLS support (default off) */ #disable_tls = yes disable_tls = no listen = tls:xx.xx.x.xx.x tls_verify_server = 1 tls_verify_client = 1 tls_require_client_certificate = 0 tls_method = TLSv1 #tls_certificate = "/usr/local/eyeball/license/cert.pem" #tls_private_key = "/usr/local/eyeball/license/privkey.pem" #tls_ca_list = "/usr/local/eyeball/license/splendor3.crtpvk.pem" tls_certificate = "/usr/local/etc/openser/tls/user/user-cert.pem" tls_private_key = "/usr/local/etc/openser/tls/user/user-privkey.pem" tls_ca_list = "/usr/local/etc/openser/tls/user/user-calist.pem" #port=5060 /* uncomment and configure the following line if you want openser to bind on a specific interface/port/proto (default bind on all available) */ listen=udp:87.236.144.13:5060 #listen=tcp:87.236.144.12:5060 sip_warning=yes

2 4

Re: [OpenSER-Users] OpenSER sending 483: Too many Hops
by Bogdan-Andrei Iancu 26 Mar '08

26 Mar '08

Marcio, you need the following line: alias=""spo.voip.br" Right now you have only: alias=""vmw02-src.spo.voip.br" Regards, Bogdan Marcio wrote: > Thanks Bogdan..but didn't work....I set the > "alias"(alias="vmw02-src.spo.voip.br <http://vmw02-src.spo.voip.br> > <http://vmw02-src.spo.voip.br>" into "openser.cfg" and it`s > ok...passed in "vmw02-srv~#openser -c" but with the parameter "listen" > didn't accept....(listen=udp:192.168.0.10:5060 > <http://192.168.0.10:5060> < http://192.168.0.10:5060>) > > regards > Marcio > > > 2008/3/25, Bogdan-Andrei Iancu <bogdan(a)voice-system.ro > <mailto:bogdan@voice-system.ro>>: > > Hi Marcio, > > Try setting spo.voip.br <http://spo.voip.br> <http://spo.voip.br> > as alias in your cfg. > > Regards, > Bogdan > > Marcio wrote: > > Hi, > > > > If anybody can take a look openser.cfg below. I`m working vmware > > (enviroments virtual)...Whenever I try a "register" a client I get > > 483: Too many Hops....as if he had a loop (config file) but I am > > unable to find the problem... > > > > I use OpenSER 1.3-notls, client softphone: twincle or xlite... > > > > OpenSER - 192.168.0.10 <http://192.168.0.10> <http://192.168.0.10> > > Client - 192.168.0.25 <http://192.168.0.25> <http://192.168.0.25> > > > > Bind9/named.conf(SRV, A): spo.voip.br <http://spo.voip.br> > <http://spo.voip.br> > > > > > > Does anyone know how to fix this ? > > > > Any hints.....?? > > > > thx... > > > > > > > > *OpenSER File:* > > ** > > > > ####### Global Parameters ######### > > > > alias="vmw02-srv.spo.voip.br <http://vmw02-srv.spo.voip.br> > <http://vmw02-srv.spo.voip.br>" > > > > debug=6 > > > > log_stderror=no > > > > log_facility=LOG_LOCAL0 > > > > log_name="Openser" > > > > fork=yes > > > > children=8 > > > > tcp_children=4 > > > > group="openser" > > > > user="openser" > > > > /* uncomment the following lines to enable debugging */ > > > > #debug=6 > > > > #fork=no > > > > #log_stderror=yes > > > > /* uncomment the next line to disable TCP (default on) */ > > > > disable_tcp=yes > > > > /* uncomment the next line to enable the auto temporary > blacklisting of > > > > not available destinations (default disabled) */ > > > > #disable_dns_blacklist=no > > > > /* uncomment the next line to enable IPv6 lookup after IPv4 dns > > > > lookup failures (default disabled) */ > > > > #dns_try_ipv6=yes > > > > /* uncomment the next line to disable the auto discovery of > local aliases > > > > based on revers DNS on IPs (default on) */ > > > > #auto_aliases=no > > > > /* uncomment the following lines to enable TLS support (default > off) */ > > > > #disable_tls = no > > > > #listen = tls:your_IP:5061 > > > > #tls_verify_server = 1 > > > > #tls_verify_client = 1 > > > > #tls_require_client_certificate = 0 > > > > #tls_method = TLSv1 > > > > #tls_certificate = "/usr/local/etc/openser/tls/user/user-cert.pem" > > > > #tls_private_key = > "/usr/local/etc/openser/tls/user/user-privkey.pem" > > > > #tls_ca_list = "/usr/local/etc/openser/tls/user/user-calist.pem" > > > > > > > > port=5060 > > > > /* uncomment and configure the following line if you want openser to > > > > bind on a specific interface/port/proto (default bind on all > available) */ > > > > #listen=udp:192.168.1.2:5060 <http://192.168.1.2:5060> > <http://192.168.1.2:5060> > > > > listen=udp:192.168.0.10:5060 <http://192.168.0.10:5060> > <http://192.168.0.10:5060> > > > > > > > > ####### Modules Section ######## > > > > #set module path > > > > mpath="/usr/local/lib/openser/modules/" > > > > /* uncomment next line for MySQL DB support */ > > > > loadmodule "mysql.so" > > > > loadmodule "sl.so" > > > > loadmodule "tm.so" > > > > loadmodule "rr.so" > > > > loadmodule "maxfwd.so" > > > > modparam("maxfwd", "max_limit", 256) > > > > loadmodule "usrloc.so" > > > > loadmodule "registrar.so" > > > > loadmodule "textops.so" > > > > loadmodule "mi_fifo.so" > > > > loadmodule "uri_db.so" > > > > loadmodule "uri.so" > > > > loadmodule "xlog.so" > > > > loadmodule "acc.so" > > > > loadmodule "avpops.so" > > > > /* uncomment next lines for MySQL based authentication support > > > > NOTE: a DB (like mysql) module must be also loaded */ > > > > loadmodule "auth.so" > > > > loadmodule "auth_db.so" > > > > /* uncomment next line for aliases support > > > > NOTE: a DB (like mysql) module must be also loaded */ > > > > loadmodule "alias_db.so" > > > > /* uncomment next line for multi-domain support > > > > NOTE: a DB (like mysql) module must be also loaded > > > > NOTE: be sure and enable multi-domain support in all used modules > > > > (see "multi-module params" section ) */ > > > > loadmodule "domain.so" > > > > /* uncomment the next two lines for presence server support > > > > NOTE: a DB (like mysql) module must be also loaded */ > > > > #loadmodule "presence.so" > > > > #loadmodule "presence_xml.so" > > > > > > > > # ----------------- setting module-specific parameters > --------------- > > > > > > > > # ----- mi_fifo params ----- > > > > modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") > > > > > > > > # ----- rr params ----- > > > > # add value to ;lr param to cope with most of the UAs > > > > modparam("rr", "enable_full_lr", 1) > > > > # do not append from tag to the RR (no need for this script) > > > > #modparam("rr", "append_fromtag", 0) > > > > > > > > # ----- rr params ----- > > > > #modparam("registrar", "method_filtering", 1) > > > > /* uncomment the next line to disable parallel forking via > location */ > > > > # modparam("registrar", "append_branches", 0) > > > > /* uncomment the next line not to allow more than 10 contacts > per AOR */ > > > > #modparam("registrar", "max_contacts", 10) > > > > > > > > # ----- uri_db params ----- > > > > /* by default we disable the DB support in the module as we do > not need it > > > > in this configuration */ > > > > modparam("uri_db", "use_uri_table", 0) > > > > modparam("uri_db", "db_url", "") > > > > > > > > # ----- acc params ----- > > > > /* what sepcial events should be accounted ? */ > > > > modparam("acc", "early_media", 1) > > > > modparam("acc", "report_ack", 1) > > > > modparam("acc", "report_cancels", 1) > > > > /* by default ww do not adjust the direct of the sequential > requests. > > > > if you enable this parameter, be sure the enable "append_fromtag" > > > > in "rr" module */ > > > > modparam("acc", "detect_direction", 0) > > > > /* account triggers (flags) */ > > > > modparam("acc", "failed_transaction_flag", 3) > > > > modparam("acc", "log_flag", 1) > > > > modparam("acc", "log_missed_flag", 2) > > > > /* uncomment the following lines to enable DB accounting also */ > > > > modparam("acc", "db_flag", 1) > > > > modparam("acc", "db_missed_flag", 2) > > > > > > > > # ----- usrloc params ----- > > > > #modparam("usrloc", "db_mode", 0) > > > > /* uncomment the following lines if you want to enable DB > persistency > > > > for location entries */ > > > > modparam("usrloc", "db_mode", 2) > > > > modparam("usrloc", "db_url", > > "mysql://openser:openserrw@192.168.0.10/openser") > > > > modparam("usrloc", "user_column", "username") > > > > modparam("usrloc", "domain_column", "domain") > > > > modparam("usrloc", "contact_column", "contact") > > > > modparam("usrloc", "expires_column", "expires") > > > > modparam("usrloc", "q_column", "q") > > > > modparam("usrloc", "callid_column", "callid") > > > > modparam("usrloc", "cseq_column", "cseq") > > > > modparam("usrloc", "methods_column", "methods") > > > > modparam("usrloc", "flags_column", "flags") > > > > modparam("usrloc", "user_agent_column", "user_agent") > > > > modparam("usrloc", "received_column", "received") > > > > modparam("usrloc", "socket_column", "socket") > > > > modparam("usrloc", "use_domain", 0) > > > > modparam("usrloc", "desc_time_order", 0) > > > > modparam("usrloc", "timer_interval", 60) > > > > modparam("usrloc", "matching_mode", 0) > > > > modparam("usrloc", "cseq_delay", 20) > > > > # ----- auth_db params ----- > > > > /* uncomment the following lines if you want to enable the DB based > > > > authentication */ > > > > #modparam("auth_db", "calculate_ha1", yes) > > > > #modparam("auth_db", "password_column", "password") > > > > modparam("auth_db", "db_url", > > "mysql://openser:openserrw@192.168.0.10/openser") > > > > #modparam("auth_db", "load_credentials", "") > > > > > > > > # ----- alias_db params ----- > > > > /* uncomment the following lines if you want to enable the DB based > > > > aliases */ > > > > modparam("alias_db", "db_url", > > "mysql://openser:openserrw@192.168.0.10/openser") > > > > > > > > # ----- domain params ----- > > > > /* uncomment the following lines to enable multi-domain detection > > > > support */ > > > > modparam("domain", "db_url", > > "mysql://openser:openserrw@192.168.0.10/openser") > > > > modparam("domain", "db_mode", 1) # Use caching > > > > # ----- multi-module params ----- > > > > /* uncomment the following line if you want to enable > multi-domain support > > > > in the modules (dafault off) */ > > > > modparam("alias_db|auth_db|usrloc|uri_db", "use_domain", 1) > > > > > > > > # ----- presence params ----- > > > > /* uncomment the following lines if you want to enable presence */ > > > > #modparam("presence|presence_xml", "db_url", > > > > # "mysql://openser:openserrw@192.168.1.3/openser_1_3") > > > > #modparam("presence_xml", "force_active", 1) > > > > #modparam("presence", "server_address", "sip:192.168.1.2:5060 > <http://192.168.1.2:5060> > > <http://192.168.1.2:5060>") > > > > > > > > ####### Routing Logic ######## > > > > > > > > # main request routing logic > > > > route{ > > > > if (!mf_process_maxfwd_header("10")) { > > > > sl_send_reply("483","Too Many Hops"); > > > > exit; > > > > } > > > > if (has_totag()) { > > > > # sequential request withing a dialog should > > > > # take the path determined by record-routing > > > > if (loose_route()) { > > > > if (is_method("BYE")) { > > > > setflag(1); # do accouting ... > > > > setflag(3); # ... even if the transaction fails > > > > } > > > > route(1); > > > > } else { > > > > sl_send_reply("404","Not here"); > > > > } > > > > exit; > > > > } > > > > #initial requests > > > > # CANCEL processing > > > > if (is_method("CANCEL")) > > > > { > > > > if (t_check_trans()) > > > > t_relay(); > > > > exit; > > > > } > > > > t_check_trans(); > > > > # DESCOMENTEI O BLOCO IF{} > > > > # authenticate if from local subscriber (uncomment to enable auth) > > > > if (!(method=="REGISTER") && from_uri==myself) > > > > { > > > > if (!proxy_authorize("", "subscriber")) { > > > > proxy_challenge("", "0"); > > > > exit; > > > > } > > > > if (!check_from()) { > > > > sl_send_reply("403","Forbidden auth ID"); > > > > exit; > > > > } > > > > consume_credentials(); > > > > # caller authenticated > > > > } > > > > # record routing > > > > if (!is_method("REGISTER|MESSAGE")) > > > > record_route(); > > > > # account only INVITEs > > > > if (is_method("INVITE")) { > > > > setflag(1); # do accouting > > > > } > > > > if (!uri==myself) > > > > /* replace with following line if multi-domain support is used */ > > > > ##if (!is_uri_host_local()) > > > > { > > > > append_hf("P-hint: outbound\r\n"); > > > > # if you have some interdomain connections via TLS > > > > ##if($rd=="tls_domain1.net") { > > > > ## t_relay("tls:domain1.net <http://domain1.net> > <http://domain1.net>"); > > > > ## exit; > > > > ##} else if($rd=="tls_domain2.net") { > > > > ## t_relay("tls:domain2.net <http://domain2.net> > <http://domain2.net>"); > > > > ## exit; > > > > ##} > > > > route(1); > > > > } > > > > # requests for my domain > > > > /* uncomment this if you want to enable presence server > > > > and comment the next 'if' block > > > > NOTE: uncomment also the definition of route[2] from below */ > > > > ##if( is_method("PUBLISH|SUBSCRIBE")) > > > > ## route(2); > > > > if (is_method("PUBLISH")) > > > > { > > > > sl_send_reply("503", "Service Unavailable"); > > > > exit; > > > > } > > > > if (is_method("REGISTER")) > > > > { > > > > # ACRESCENTEI O IP ADDR - MRG > > > > # authenticate the REGISTER requests (uncomment to enable auth) > > > > #if (!www_authorize("vmw02-srv", "subscriber")) > > > > #{ > > > > # www_challenge("vmw02-srv", "0"); > > > > # exit; > > > > #} > > > > # > > > > #if (!check_to()) > > > > ##{ > > > > ## sl_send_reply("403","Forbidden auth ID"); > > > > ## exit; > > > > ##} > > > > if (!save("location")) > > > > sl_reply_error(); > > > > exit; > > > > } > > > > if ($rU==NULL) { > > > > # request with no Username in RURI > > > > sl_send_reply("484","Address Incomplete"); > > > > exit; > > > > } > > > > # apply DB based aliases (uncomment to enable) > > > > alias_db_lookup("dbaliases"); > > > > if (!lookup("location")) { > > > > switch ($retcode) { > > > > case -1: > > > > case -3: > > > > t_newtran(); > > > > t_reply("404", "Not Found"); > > > > exit; > > > > case -2: > > > > sl_send_reply("405", "Method Not Allowed"); > > > > exit; > > > > } > > > > } > > > > # when routing via usrloc, log the missed calls also > > > > setflag(2); > > > > route(1); > > > > } > > > > > > > > route[1] { > > > > # for INVITEs enable some additional helper routes > > > > if (is_method("INVITE")) { > > > > t_on_branch("2"); > > > > t_on_reply("2"); > > > > t_on_failure("1"); > > > > } > > > > if (!t_relay()) { > > > > sl_reply_error(); > > > > }; > > > > exit; > > > > } > > > > > > > > # Presence route > > > > /* uncomment the whole following route for enabling presence > > > > NOTE: do not forget to enable the call of this route from the main > > > > route */ > > > > ##route[2] > > > > ##{ > > > > ## if (!t_newtran()) > > > > ## { > > > > ## sl_reply_error(); > > > > ## exit; > > > > ## }; > > > > ## > > > > ## if(is_method("PUBLISH")) > > > > ## { > > > > ## handle_publish(); > > > > ## t_release(); > > > > ## } > > > > ## else > > > > ## if( is_method("SUBSCRIBE")) > > > > ## { > > > > ## handle_subscribe(); > > > > ## t_release(); > > > > ## } > > > > ## > > > > ## exit; > > > > ##} > > > > > > > > branch_route[2] { > > > > xlog("new branch at $ru\n"); > > > > } > > > > > > > > onreply_route[2] { > > > > xlog("incoming reply\n"); > > > > } > > > > > > > > failure_route[1] { > > > > if (t_was_cancelled()) { > > > > exit; > > > > } > > > > # uncomment the following lines if you want to block client > > > > # redirect based on 3xx replies. > > > > ##if (t_check_status("3[0-9][0-9]")) { > > > > ##t_reply("404","Not found"); > > > > ## exit; > > > > ##} > > > > # uncomment the following lines if you want to redirect the failed > > > > # calls to a different new destination > > > > ##if (t_check_status("486|408")) { > > > > ## sethostport("192.168.2.100:5060 <http://192.168.2.100:5060> > <http://192.168.2.100:5060>"); > > > > ## append_branch(); > > > > ## # do not set the missed call flag again > > > > ## t_relay(); > > > > ##} > > > > } > > > > > > > > *Ngrep -d eth1 -qt -W byline port 5060* > > > > root@vmw02-srv:~# ngrep -d eth1 -qt -W byline port 5060 > > > > interface: eth1 (192.168.0.0/255.255.255.0 > <http://192.168.0.0/255.255.255.0> > > <http://192.168.0.0/255.255.255.0>) > > > > filter: (ip or ip6) and ( port 5060 ) > > > > U 2008/03/20 13:57:31.763060 192.168.0.25:5060 > <http://192.168.0.25:5060> > > <http://192.168.0.25:5060> -> 192.168.0.10:5060 > <http://192.168.0.10:5060> <http://192.168.0.10:5060> > > > > REGISTER sip:spo.voip.br <http://spo.voip.br> > <http://spo.voip.br> SIP/2.0. > > > > Via: SIP/2.0/UDP 192.168.0.25 <http://192.168.0.25> > > <http://192.168.0.25>;rport;branch=z9hG4bKignjupsg. > > > > Max-Forwards: 70. > > > > To: "vmwcli03" <sip:vmwcli03@spo.voip.br > <mailto:sip:vmwcli03@spo.voip.br> > > <mailto:sip <mailto:sip>:vmwcli03@spo.voip.br > <mailto:vmwcli03@spo.voip.br>>>. > > > > From: "vmwcli03" <sip:vmwcli03@spo.voip.br > <mailto:sip:vmwcli03@spo.voip.br> > > <mailto:sip <mailto:sip>:vmwcli03@spo.voip.br > <mailto:vmwcli03@spo.voip.br>>>;tag=dcoan. > > > > Call-ID: iswtwciijecswfp(a)192.168.0.25 > <mailto:iswtwciijecswfp@192.168.0.25> > > <mailto:iswtwciijecswfp@192.168.0.25 > <mailto:iswtwciijecswfp@192.168.0.25>>. > > > > CSeq: 883 REGISTER. > > > > Contact: <sip:vmwcli03@192.168.0.25 > <mailto:sip:vmwcli03@192.168.0.25> > > <mailto:sip <mailto:sip>:vmwcli03@192.168.0.25 > <mailto:vmwcli03@192.168.0.25>>>;expires=3600. > > > > Allow: > INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO. > > > > User-Agent: Twinkle/1.0. > > > > Content-Length: 0. > > > > . > > > > > > > > U 2008/03/20 13:57:32.333159 192.168.0.25:5060 > <http://192.168.0.25:5060> > > <http://192.168.0.25:5060> -> 192.168.0.10:5060 > <http://192.168.0.10:5060> <http://192.168.0.10:5060> > > > > REGISTER sip:spo.voip.br <http://spo.voip.br> > <http://spo.voip.br> SIP/2.0. > > > > Via: SIP/2.0/UDP 192.168.0.25 <http://192.168.0.25> > > <http://192.168.0.25>;rport;branch=z9hG4bKignjupsg. > > > > Max-Forwards: 70. > > > > To: "vmwcli03" <sip:vmwcli03@spo.voip.br > <mailto:sip:vmwcli03@spo.voip.br> > > <mailto:sip <mailto:sip>:vmwcli03@spo.voip.br > <mailto:vmwcli03@spo.voip.br>>>. > > > > From: "vmwcli03" <sip:vmwcli03@spo.voip.br > <mailto:sip:vmwcli03@spo.voip.br> > > <mailto:sip <mailto:sip>:vmwcli03@spo.voip.br > <mailto:vmwcli03@spo.voip.br>>>;tag=dcoan. > > > > Call-ID: iswtwciijecswfp(a)192.168.0.25 > <mailto:iswtwciijecswfp@192.168.0.25> > > <mailto:iswtwciijecswfp@192.168.0.25 > <mailto:iswtwciijecswfp@192.168.0.25>>. > > > > CSeq: 883 REGISTER. > > > > Contact: <sip:vmwcli03@192.168.0.25 > <mailto:sip:vmwcli03@192.168.0.25> > > <mailto:sip <mailto:sip>:vmwcli03@192.168.0.25 > <mailto:vmwcli03@192.168.0.25>>>;expires=3600. > > > > Allow: > INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO. > > > > User-Agent: Twinkle/1.0. > > > > Content-Length: 0. > > > > . > > > > > > > > U 2008/03/20 13:57:33.321952 192.168.0.25:5060 > <http://192.168.0.25:5060> > > <http://192.168.0.25:5060> -> 192.168.0.10:5060 > <http://192.168.0.10:5060> <http://192.168.0.10:5060> > > > > REGISTER sip:spo.voip.br <http://spo.voip.br> > <http://spo.voip.br> SIP/2.0. > > > > Via: SIP/2.0/UDP 192.168.0.25 <http://192.168.0.25> > > <http://192.168.0.25>;rport;branch=z9hG4bKignjupsg. > > > > Max-Forwards: 70. > > > > To: "vmwcli03" <sip:vmwcli03@spo.voip.br > <mailto:sip:vmwcli03@spo.voip.br> > > <mailto:sip <mailto:sip>:vmwcli03@spo.voip.br > <mailto:vmwcli03@spo.voip.br>>>. > > > > From: "vmwcli03" <sip:vmwcli03@spo.voip.br > <mailto:sip:vmwcli03@spo.voip.br> > > <mailto:sip <mailto:sip>:vmwcli03@spo.voip.br > <mailto:vmwcli03@spo.voip.br>>>;tag=dcoan. > > > > Call-ID: iswtwciijecswfp(a)192.168.0.25 > <mailto:iswtwciijecswfp@192.168.0.25> > > <mailto:iswtwciijecswfp@192.168.0.25 > <mailto:iswtwciijecswfp@192.168.0.25>>. > > > > CSeq: 883 REGISTER. > > > > Contact: <sip:vmwcli03@192.168.0.25 > <mailto:sip:vmwcli03@192.168.0.25> > > <mailto:sip <mailto:sip>:vmwcli03@192.168.0.25 > <mailto:vmwcli03@192.168.0.25>>>;expires=3600. > > > > Allow: > INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO. > > > > User-Agent: Twinkle/1.0. > > > > Content-Length: 0. > > > > . > > > > > > > > *U 2008/03/20 13:57:35.309584 192.168.0.25:5060 > <http://192.168.0.25:5060> > > <http://192.168.0.25:5060> -> 192.168.0.10:5060 > <http://192.168.0.10:5060> > > <http://192.168.0.10:5060>* > > > > *REGISTER sip:spo.voip.br <http://spo.voip.br> > <http://spo.voip.br> SIP/2.0.* > > > > Via: SIP/2.0/UDP 192.168.0.25 <http://192.168.0.25> > > <http://192.168.0.25>;rport;branch=z9hG4bKignjupsg. > > > > Max-Forwards: 70. > > > > To: "vmwcli03" <sip:vmwcli03@spo.voip.br > <mailto:sip:vmwcli03@spo.voip.br> > > <mailto:sip <mailto:sip>:vmwcli03@spo.voip.br > <mailto:vmwcli03@spo.voip.br>>>. > > > > From: "vmwcli03" <sip:vmwcli03@spo.voip.br > <mailto:sip:vmwcli03@spo.voip.br> > > <mailto:sip <mailto:sip>:vmwcli03@spo.voip.br > <mailto:vmwcli03@spo.voip.br>>>;tag=dcoan. > > > > Call-ID: iswtwciijecswfp(a)192.168.0.25 > <mailto:iswtwciijecswfp@192.168.0.25> > > <mailto:iswtwciijecswfp@192.168.0.25 > <mailto:iswtwciijecswfp@192.168.0.25>>. > > > > CSeq: 883 REGISTER. > > > > Contact: <sip:vmwcli03@192.168.0.25 > <mailto:sip:vmwcli03@192.168.0.25> > > <mailto:sip <mailto:sip>:vmwcli03@192.168.0.25 > <mailto:vmwcli03@192.168.0.25>>>;expires=3600. > > > > Allow: > INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO. > > > > User-Agent: Twinkle/1.0. > > > > Content-Length: 0. > > > > . > > > > > > > > *U 2008/03/20 13:57:37.026703 192.168.0.10:5060 > <http://192.168.0.10:5060> > > <http://192.168.0.10:5060> -> 192.168.0.25:5060 > <http://192.168.0.25:5060> > > <http://192.168.0.25:5060>* > > > > *SIP/2.0 483 Too Many Hops.* > > > > Via: SIP/2.0/UDP 192.168.0.25 <http://192.168.0.25> > > <http://192.168.0.25>;rport=5060;branch=z9hG4bKignjupsg. > > > > To: "vmwcli03" <sip:vmwcli03@spo.voip.br > <mailto:sip:vmwcli03@spo.voip.br> > > <mailto:sip <mailto:sip>:vmwcli03@spo.voip.br > <mailto:vmwcli03@spo.voip.br>>>;tag=1333af5435552d3677a0ca954c3a6e9f.6889. > > > > From: "vmwcli03" <sip:vmwcli03@spo.voip.br > <mailto:sip:vmwcli03@spo.voip.br> > > <mailto:sip <mailto:sip>:vmwcli03@spo.voip.br > <mailto:vmwcli03@spo.voip.br>>>;tag=dcoan. > > > > Call-ID: iswtwciijecswfp(a)192.168.0.25 > <mailto:iswtwciijecswfp@192.168.0.25> > > <mailto:iswtwciijecswfp@192.168.0.25 > <mailto:iswtwciijecswfp@192.168.0.25>>. > > > > CSeq: 883 REGISTER. > > > > Server: OpenSER (1.3.0-notls (i386/linux)). > > > > Content-Length: 0. > > > > > > > > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Users mailing list > > Users(a)lists.openser.org <mailto:Users@lists.openser.org> > > http://lists.openser.org/cgi-bin/mailman/listinfo/users > >

1 0

Re: [OpenSER-Users] Info about redirect function
by Bogdan-Andrei Iancu 26 Mar '08

26 Mar '08

Hi Valerio, not sure if this will be sufficient as, as far I understood, you take care only on signalling part. You should also look for RTP stream. Regards, Bogdan Valerio Di Marino wrote: > Hi Andrei, > thanks for your answer: I already seen the file redirect.cfg. > > My doubt is about the possibility to solve the problem with NAT. > > Using a public proxy I need to open several ports on the canopy router > (see my first message): in other case I can't listen the caller. > Mi idea is to install a redirect server in my network, on a my public > machine, pointing my ATA on this: redirect server will redirect data > to the public proxy server. Is it possible to do this, and should this > configuration solve my problem ? > > In this moment, I cant try this solution... > > Thanks > Valerio > > 2008/3/25, Bogdan-Andrei Iancu <bogdan(a)voice-system.ro > <mailto:bogdan@voice-system.ro>>: > > Hi Valerio, > > If you want to learn how to do redirect with openser, take a look at: > examples/redirect.cfg > > in the SVN tree. > > Regards, > Bogdan > > > Valerio Di Marino wrote: > > Good morning to all. > > > > I am Valerio, and this im my first post. > > I was never using OPENSER: I discovered it this morning, > searcing for > > a solution. > > > > I have this situation: I am using Motorola Canopy wireless > antennas to > > deploy internet connection, but it implements a symmetric NAT which > > make not operating STUN protocol. > > I think that one of solution should be to install a proxy server > which > > redirects registering and calls to the public proxy (euteliavoip). > > I thought that OPENSER should be a good solution: can you > confirm this? > > Thanks for every your suggestions. > > Valerio > > > > > > > > > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Users mailing list > > Users(a)lists.openser.org <mailto:Users@lists.openser.org> > > http://lists.openser.org/cgi-bin/mailman/listinfo/users > > > >

1 0

Re: [OpenSER-Users] Carrierroute
by Douglas Garstang 26 Mar '08

26 Mar '08

> > >I am using this module in production with more then 400k routes loaded >for 8 different providers and it is working like a charm. >As Henning mentioned, the interface is a little bit cumbersome but usable. >You can put all the carriers under the same table and prioritize the >gateway based on domains. Like tis you don't need to hardcode >anything in the config file. > >Carrierroute was designed to deal with a big amount of routing rules >and it is doing an amazing job. >LCR was designed to deal with a limited number of routes/gateways but >t has a different level of flexibility and a more mature interface. Ovidiu, ok... but I'm still a bit confused. How do you simulate LCR with carrierroute? Doug. ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ

1 0

Re: [OpenSER-Users] Carrierroute
by Douglas Garstang 26 Mar '08

26 Mar '08

> > >On Tue, March 25, 2008 3:24 am, Ovidiu Sas wrote: >> The database config is right in the README file. >> >> Here are the trees: >> [..] >> I agree with you that the names are confusing :) >> You may or may not need to use the subscriber tables, based on what >> functions are you using. >> In the example that I gave you, you don't need the location table. >> Instead, you force a routing tree (in my example is the "default" >> tree). > >Hi, > >yes, they are probably still a few things in the whole module terminology >that are not adequate explained, even if the documentation has already >seen quite a lot of work. As this module can be used to implement pretty >complex scenarios, the documentation shows this too, unfortunately. Complex scenarios? To change the carrier mid-call, you need to hard code it. Why isn't there a function which retrieves the next carrier from the database, rather than having to hard code it? There is also NO lcr at all. We absolutely need lcr. It also seems you are limited to a maximum of two carriers? > >You need only data from a user specific table, e.g. subscriber if you need >to implement user specific routing. e.g. choose carrier A for one user >set, carrier B for another one.. If you don't need this, then you not need >to load data from this table. What about multiple carriers per user with LCR and multiple gateways? > >You have a list of carriers in your routing space, and each of them can >have again a list of domains. As every domain can route also to many >different prefixes (with several hosts per prefix), this module refers to >them as "trees". A list of carriers is fine, but how do I switch carriers without hard coding the carrier in the cr_tree_rewrite_uri() function? Using a simple example, using carrierroute, if we where calling say prefix 1303, we'd have to call cr_tree_rewrite_uri('Verizon'), and then if Verizon fails, and we do failover, call cr_tree_rewrite_uri('Teleglobe'). See the problem here? The carriers are hard coded in the openser.cfg file, and there's no LCR. What am I missing? It can't be that horrible. > >If you want to implement simple LCR, then you don't need the functions >that allows you to choose a carrier. Again, i agree that the actual >interface in the stable branch is a little bit cumbersome to use, the new >interface currently in development will be hopefully address some of this >issues. The LCR module seems to suit our need better, but the number of routes is hard coded in the code to 256. Why is that? We use several carriers, and each has THOUSANDS of routes. Verizon for example has about 10,000 routes and Teleglobe has about 7,000 routes. Why is the limit so low? I increased the limit in lcr_mod.c, and increased the size of the private and shared memory, which allowed the lcr table to load, but what are the correct values? How do I know that it won't run out of memory? If that happens, do I just increase the private and shared memory? Doug ____________________________________________________________________________________ Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping

2 1

Re: [OpenSER-Users] Carrierroute
by Douglas Garstang 25 Mar '08

25 Mar '08

Ovidiu, I need database config info.... your example uses the config file. What is a 'tree'? The documentation keeps talking about it, but I don'tsee it defined anywhere. Carrierroute SEEMS to need the subscribertable, but it doesn't load it from the database.... why not? Why should I spend days experimenting to get it to work? If the docs where adequate, that wouldn't be required. Doug. ----- Original Message ---- From: Ovidiu Sas <osas(a)voipembedded.com> To: Douglas Garstang <dougmig33(a)yahoo.com> Cc: users(a)lists.openser.org Sent: Monday, March 24, 2008 5:43:53 PM Subject: Re: [OpenSER-Users] Carrierroute Hello Douglas, The database sample is available in the README file here: http://www.openser.org/docs/modules/1.3.x/carrierroute.html#AEN575 The entire documentation is there, you just need to read it carefully and experiment. You had a similar post about the documentation for the lcr module and in the end it seems that you figure it out :) If you think that you can improve documentation, feel free to send patches for the README file. Here you can find a configuration file example for carrierroute: http://voipembedded.com/resources/openser_cr.cfg Regards, Ovidiu Sas On Mon, Mar 24, 2008 at 7:52 PM, Douglas Garstang <dougmig33(a)yahoo.com> wrote: > > I have been trying to get the carrierroute module to work ALL day. > Can someone 'supplement' the openser docs with maybe some sample database > data? > > The docs really aren't making sense. The docs keep talking about a routing > tree? What is a routing tree? > When I start openser, I don't see it query the subscriber table. Should it? > > Thanks, > Doug. > > > __________________________________________________ > Do You Yahoo!? > Tired of spam? Yahoo! Mail has the best spam protection around > http://mail.yahoo.com > _______________________________________________ > Users mailing list > Users(a)lists.openser.org > http://lists.openser.org/cgi-bin/mailman/listinfo/users > > ____________________________________________________________________________________ Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs

3 2

[OpenSER-Users] CFP - IEEE Workshop NGNOPS08 - Paper Submission until 14 April 2008
by Ramona Modroiu 25 Mar '08

25 Mar '08

Apology for possible multiple copies due to cross posting ---------------------------------- First International Workshop on Next Generation Networks: Open Platforms and Services (NGNOPS'08) will be held in Cardiff, UK this September. The NGNOPS'08 workshop is colocated with the IEEE NGMAST'08 conference (http://www.comp.glam.ac.uk/NGMAST08/) and scheduled on the Sep 18, 2008. NGNOPS'08 (http://www.comp.glam.ac.uk/NGMAST08/workshops/ngnops/index.htm) is aiming at a convergence of Open Source community with the academic community and industry players related to next-generation networking, technologies, applications and services. The NGNOPS'08 program will comprise of paper presentations and a couple of keynote sessions. Paper submission and more details are available at http://www.comp.glam.ac.uk/ngmast08/workshops/ngnops/index.htm. For any questions, you can write me an email. Best regards, Ramona Modroiu

1 0

[OpenSER-Users] Wrong R-uri in ACK
by Lucio Maciel 25 Mar '08

25 Mar '08

Hi... I'm getting some wrongs ruri's in the ACK for a final 200ok from an INVITE... It is comming as the original INVITE r-uri (number@host) and not the contact from the 200ok. There is something i can do to workaround this in openser? Thanks. -- ------------------------- - Lúcio Fernando Maciel - - lucio(a)tesatelecom.com - - Tesa Telecom - - R. Augusto Meyer 163 - - sala 603/604 - - +55 (51) 30615569 - - +55 (51) 84733802 - -------------------------

3 7

[OpenSER-Users] LCR... Too many entries
by Douglas Garstang 25 Mar '08

25 Mar '08

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? Mar 25 17:44:49 s_local@st008/St008 /usr/local/sbin/openser[7517]: INFO:xlog:mod_init: initializing... Mar 25 17:44:49 s_local@st008/St008 /usr/local/sbin/openser[7517]: ERROR:lcr:reload_gws: Too many lcr entries <4999> Mar 25 17:44:49 s_local@st008/St008 /usr/local/sbin/openser[7517]: CRITICAL:lcr:mod_init: Failed to reload gateways and routes Mar 25 17:44:49 s_local@st008/St008 /usr/local/sbin/openser[7517]: ERROR:core:init_mod: failed to initialize module lcr Mar 25 17:44:49 s_local@st008/St008 /usr/local/sbin/openser[7517]: ERROR:core:main: error while initializing modules 5000 LCR entries is NOT excessive. We have almost 10,000 routes from Verizon alone. Doug. ____________________________________________________________________________________ Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs

2 1

[OpenSER-Users] uac_replace_from and CANCEL
by Andreas Granig 25 Mar '08

25 Mar '08

Hi, I hit another problem with a Cisco PGW in combination with CANCEL, and I'm not sure which fault it is. If I do uac_replace_from in the INVITE, the From-Header is altered somehow and sent to the PGW. So if A calls B where B is a PGW, the From from A to OpenSER is for example sip:a@somedomain and from OpenSER to B it's sip:other-a@somedomain. If the call is cancelled, the From-header isn't altered, so the From from A to OpenSER is sip:a@somedomain and from OpenSER to B it's sip:a@somedomain as well. Now the PGW seems to ignore this CANCEL, and I guess it's because of the different From-usernames, since it works if I don't do any uac_replace_from. RFC3261 says in §9.1: The Request-URI, Call-ID, To, the numeric part of CSeq, and From header fields in the CANCEL request MUST be identical to those in the request being cancelled, including tags. So is it correct behavior of the PGW because the From header in the CANCEL is different from the From header in the INVITE? Should OpenSER alter the From in the CANCEL as well? Or should the PGW just check the From tags, but not the From URI? Andreas

5 23

[OpenSER-Users] new CDRTool release 6.4.0
by Adrian Georgescu 25 Mar '08

25 Mar '08

The software can be downloaded from: http://download.dns-hosting.info/CDRTool/ Changelog since 6.3.0: cdrtool (6.4.0) unstable; urgency=high * Refactor quota system to work with millions of sip accounts * Removed memcache usage for quota * Fixed voicemail divert setting * Separate unblock quota from init quota operation Added domain and notified columns to quota_usage Show quota_usage table in rating section * Fixed syncronization of quota and blocked by quota * Use normalized lock during init of quota to avoid race conditions * Added quota_usage table Must apply mysql changes from setup/mysql/alter_tables.mysql * Allow for empty prefixes in PSTN routes * Fixes in update of impersonate field cdrtool (6.3.2) unstable; urgency=low * Log for how many account the CDRs are loaded to rebuild the quota count cdrtool (6.3.1) unstable; urgency=low * Filter CDRs for local domains when init quota counters * A subaccount cannot change his own impersonate field * Replaced _serialize function with previos pear soap version to fix empty array representation

1 0

[OpenSER-Users] Adding header to self-generated reply (sl_send_reply)
by Alex Massover 25 Mar '08

25 Mar '08

Hello! I'm looking for a way to add custom header to self-generated replies. For example I have an $avp(s:test) and want to send 4xx with header Custom: $avp(s:test). Is it possible? I didn't find a way to add headers to sl_send_reply or t_replay. -- Best Regards, Alex Massover VoIP Development Manager Jajah Inc.

2 2

Re: [OpenSER-Users] LCR Gateway Priority....
by Douglas Garstang 25 Mar '08

25 Mar '08

> >Douglas Garstang writes: > >> Thanks. I went through that option yesterday. As we have 6 instances >> of OpenSER running, I would need to have SIX routes defined in the >> lcr table for a single prefix. For example (IP's modified): > >i don't follow this. the example you give now is different from earlier >one. anyway, it is not a good idea to store 120,000 entries in lcr >table. We can't have over 120,000 lcr entries??? Your joking? Why not? They're all in memory aren't they? Providers publish rate sheets with thousands and thousands of entries, and people often use multiple providers. I'm a little confused what LCR is even for then if it can't handle this. We need 6 entries in LCR because we have 6 OpenSER systems running. Each one needs to be configured differently, to send calls to different gateways. It's as simple as that. For example, our Hong Kong OpenSER needs to send calls to the Verizon gateway in Hong Kong, while our USA OpenSER needs to send calls to the Verizon gateway in the US. They don't all go to the same place. > >> Our providers also have sometimes have several gateways in a single >> POP. However, we don't want to actually try and route calls to every >> single gateway in a POP, just a few of them. > >then you could list only those you want to try. there is no capability >to skip gateways unless you do it in your script after calling >next_gws. > >this convinces me that current pstn service model is something we would >really need to get rid of. emulating it using sip makes no sense. Don't understand you. It just seems who ever wrote the LCR module really didn't think of the real world application. Doug. ____________________________________________________________________________________ Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs

1 0

[OpenSER-Users] v1.3.1 & bla
by Jeremy George 25 Mar '08

25 Mar '08

Hello All, I had bla working with v1.3.0, but it isn't with v1.3.1 . I didn't expect a maintenance release to require a configuration change, so I am confused. V1.3.1 appears to need a pua_bla server address statement in the modparams. I wasn't using one with v1.3.0, and it appears not to be allowed. The presence server configuration is very simple, just about straight from the example. It's attached as bla.cfg I think I have an idea about what's wrong, but not why. Attached also is are ngrep traces of the first and second UAs registering correctly with v1.3.0 and the first UA registering correctly with v1.3.1 . Then an ngrep trace of the second UA registering incorrectly with v1.3.1 . The NOTIFY that should be relayed from the second UA to the first is missing. The attached files are named firstphone-register-130.log, secondphone-register-130.log, firstphone-register-131.log and secondphone-register-131.log . Any pointers to the problem would be greatly appreciated. - Jeremy

3 2

[OpenSER-Users] openser + java
by Mauricio Viveros 25 Mar '08

25 Mar '08

Hi Just one questions 1. There is some API or Framework to develop SIP Java ( JSE or JME ) Application using openser ?? Thanks, Mauricio.

2 1

Re: [OpenSER-Users] LCR Gateway Priority....
by Douglas Garstang 25 Mar '08

25 Mar '08

> >Douglas Garstang writes: > >> 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. > >it might be possible for you to achieve what you describe using existing >lcr module by having only one gw per group and then assign priorities in >lcr table in such a way that for prefix +xx highest priority is gw group >1 of the cheapest carrier, then gw group 1 of second cheapest carrier, >then gw group 2 of cheapest carrier and finally gw group 2 of second >cheapest carrier (assuming two carriers serving +xx). Juha, Thanks. I went through that option yesterday. As we have 6 instances of OpenSER running, I would need to have SIX routes defined in the lcr table for a single prefix. For example (IP's modified): mysql> select * from lcr; +----+--------+---------------+--------+----------+ | id | prefix | from_uri | grp_id | priority | +----+--------+---------------+--------+----------+ | 8 | 1303 | @203.84.213.8 | 1 | 1 | | 9 | 1303 | @203.84.214.8 | 2 | 1 | | 10 | 1303 | @203.84.215.8 | 3 | 1 | | 11 | 1303 | @203.84.216.8 | 4 | 1 | | 12 | 1303 | @203.84.217.8 | 5 | 1 | | 14 | 1303 | @203.84.218.8 | 6 | 1 | +----+--------+---------------+--------+----------+ ..... All that .... six rows for a single prefix. Of course, we have thousands of routes for each of our providers... off the top of my head, we have about 17,000 routes between Teleglobe and Verizon... multiply that by 6 and you have almost 120,000 routes. Apart from the administrative hassle, can OpenSER handle 120,000+ routes in the lcr table ok? Our providers also have sometimes have several gateways in a single POP. However, we don't want to actually try and route calls to every single gateway in a POP, just a few of them. The LCR module doesn't seem to have a way to skip to the next lcr route. Can it do that? If it can't, we have to try every single gateway in a POP before skipping to the next carrier, which just isn't feesible. Doug. -- juha ____________________________________________________________________________________ Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping

2 1

[OpenSER-Users] Problem with loading snmpstats
by Stanisław Pitucha 25 Mar '08

25 Mar '08

Hey I can't get snmpstats module to load. Message from logs is (during config parsing): ERROR:core:sr_load_module: could not open module </usr/local/openser/lib/openser/modules/snmpstats.so>: /usr/lib/libnetsnmpagent.so.10: undefined symbol: boot_DynaLoader System: CentOS-5.1, 32b, up-to-date. Openser: svn head from 1.3 branch Tried `ldconfig`, recompiling after `make clean`, ... Has anyone got some ideas how to resolve it? Thanks, Stanisław Pitucha

2 2

[OpenSER-Users] 483: Too Many Hops
by Marcio 25 Mar '08

25 Mar '08

Thanks Bogdan..but didn't work....I set the "alias"(alias=" vmw02-src.spo.voip.br <http://vmw02-src.spo.voip.br>" into "openser.cfg" and it`s ok...passed in "vmw02-srv~#openser -c" but with the parameter "listen" didn't accept....(listen=udp:192.168.0.10:5060 < http://192.168.0.10:5060>) and be receiving yet.....483:.... regards Marcio

1 0

[OpenSER-Users] Presence Server
by Stefan Schmidt 25 Mar '08

25 Mar '08

hello, iam new to this list and i have a question about the presence Server in openser. We are currently using asterisk for a hosted pbx solution with the feature so show active and also ringing lines on a snom360 phone. This feature uses the Asterisk hint, subscriber and notify handling. i havent found if i could use the presence server for this solution. The problem is that if i just pass through the subscriber and notify messages from the client via openser to the asterisk server that the server doesnt know if the client which should be watched is registered or not (registrations are made with openser). So i have to find a solution where openser could handle subscriptions for clients which are registered and also send notifys on every invite, ringing and bye of a watched client to the subscriber. could i use the presence server for that or should i search for another solution? Best regards Steve Smith -- Für weitere Fragen stehen wir gerne unter voip(a)sil.at oder 059944 - 2440 zur Verfügung. Mit freundlichen Grüssen -- Stefan Schmidt Sysadmin/VOIP // sst(a)sil.at // Tel 059944-2440// ------------------------------------------------- SILVER SERVER GmbH // Lorenz-Mandl-Gasse 33/1 // A-1160 Wien // Fax 059944-9000 // www.sil.at // -------------------------------------------------

3 4

[OpenSER-Users] "always" serial fork with LCR?
by Amit Sharma 25 Mar '08

25 Mar '08

Hi All, I am a newbie to this list so please forgive me if the question below has been discussed before. I could not find anything related so i am sending my query. I have been looking at the LCR module to do serial forking since we want to prioritize contacts based on q values. However, we do not want to fork in parallel to contacts even if they share the same q value. AFAIK,this is currently not possible with the LCR module. Would it be a good idea to have a parameter (e.g "append-branches") in the LCR module which can control the forking behavior when q value of contacts is the same? Thanks, Amit

6 11

[OpenSER-Users] OpenSER sending 483: Too many Hops
by Marcio 25 Mar '08

25 Mar '08

Hi, If anybody can take a look openser.cfg below. I`m working vmware (enviroments virtual)...Whenever I try a "register" a client I get 483: Too many Hops....as if he had a loop (config file) but I am unable to find the problem... I use OpenSER 1.3-notls, client softphone: twincle or xlite... OpenSER - 192.168.0.10 Client - 192.168.0.25 Bind9/named.conf(SRV, A): spo.voip.br Does anyone know how to fix this ? Any hints.....?? thx... *OpenSER File:* ** ####### Global Parameters ######### alias="vmw02-srv.spo.voip.br" debug=6 log_stderror=no log_facility=LOG_LOCAL0 log_name="Openser" fork=yes children=8 tcp_children=4 group="openser" user="openser" /* uncomment the following lines to enable debugging */ #debug=6 #fork=no #log_stderror=yes /* uncomment the next line to disable TCP (default on) */ disable_tcp=yes /* uncomment the next line to enable the auto temporary blacklisting of not available destinations (default disabled) */ #disable_dns_blacklist=no /* uncomment the next line to enable IPv6 lookup after IPv4 dns lookup failures (default disabled) */ #dns_try_ipv6=yes /* uncomment the next line to disable the auto discovery of local aliases based on revers DNS on IPs (default on) */ #auto_aliases=no /* uncomment the following lines to enable TLS support (default off) */ #disable_tls = no #listen = tls:your_IP:5061 #tls_verify_server = 1 #tls_verify_client = 1 #tls_require_client_certificate = 0 #tls_method = TLSv1 #tls_certificate = "/usr/local/etc/openser/tls/user/user-cert.pem" #tls_private_key = "/usr/local/etc/openser/tls/user/user-privkey.pem" #tls_ca_list = "/usr/local/etc/openser/tls/user/user-calist.pem" port=5060 /* uncomment and configure the following line if you want openser to bind on a specific interface/port/proto (default bind on all available) */ #listen=udp:192.168.1.2:5060 listen=udp:192.168.0.10:5060 ####### Modules Section ######## #set module path mpath="/usr/local/lib/openser/modules/" /* uncomment next line for MySQL DB support */ loadmodule "mysql.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" modparam("maxfwd", "max_limit", 256) loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "mi_fifo.so" loadmodule "uri_db.so" loadmodule "uri.so" loadmodule "xlog.so" loadmodule "acc.so" loadmodule "avpops.so" /* uncomment next lines for MySQL based authentication support NOTE: a DB (like mysql) module must be also loaded */ loadmodule "auth.so" loadmodule "auth_db.so" /* uncomment next line for aliases support NOTE: a DB (like mysql) module must be also loaded */ loadmodule "alias_db.so" /* uncomment next line for multi-domain support NOTE: a DB (like mysql) module must be also loaded NOTE: be sure and enable multi-domain support in all used modules (see "multi-module params" section ) */ loadmodule "domain.so" /* uncomment the next two lines for presence server support NOTE: a DB (like mysql) module must be also loaded */ #loadmodule "presence.so" #loadmodule "presence_xml.so" # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) #modparam("rr", "append_fromtag", 0) # ----- rr params ----- #modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam("registrar", "max_contacts", 10) # ----- uri_db params ----- /* by default we disable the DB support in the module as we do not need it in this configuration */ modparam("uri_db", "use_uri_table", 0) modparam("uri_db", "db_url", "") # ----- acc params ----- /* what sepcial events should be accounted ? */ modparam("acc", "early_media", 1) modparam("acc", "report_ack", 1) modparam("acc", "report_cancels", 1) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "failed_transaction_flag", 3) modparam("acc", "log_flag", 1) modparam("acc", "log_missed_flag", 2) /* uncomment the following lines to enable DB accounting also */ modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 2) # ----- usrloc params ----- #modparam("usrloc", "db_mode", 0) /* uncomment the following lines if you want to enable DB persistency for location entries */ modparam("usrloc", "db_mode", 2) modparam("usrloc", "db_url", "mysql://openser:openserrw@192.168.0.10 /openser") modparam("usrloc", "user_column", "username") modparam("usrloc", "domain_column", "domain") modparam("usrloc", "contact_column", "contact") modparam("usrloc", "expires_column", "expires") modparam("usrloc", "q_column", "q") modparam("usrloc", "callid_column", "callid") modparam("usrloc", "cseq_column", "cseq") modparam("usrloc", "methods_column", "methods") modparam("usrloc", "flags_column", "flags") modparam("usrloc", "user_agent_column", "user_agent") modparam("usrloc", "received_column", "received") modparam("usrloc", "socket_column", "socket") modparam("usrloc", "use_domain", 0) modparam("usrloc", "desc_time_order", 0) modparam("usrloc", "timer_interval", 60) modparam("usrloc", "matching_mode", 0) modparam("usrloc", "cseq_delay", 20) # ----- auth_db params ----- /* uncomment the following lines if you want to enable the DB based authentication */ #modparam("auth_db", "calculate_ha1", yes) #modparam("auth_db", "password_column", "password") modparam("auth_db", "db_url", "mysql://openser:openserrw@192.168.0.10 /openser") #modparam("auth_db", "load_credentials", "") # ----- alias_db params ----- /* uncomment the following lines if you want to enable the DB based aliases */ modparam("alias_db", "db_url", "mysql://openser:openserrw@192.168.0.10 /openser") # ----- domain params ----- /* uncomment the following lines to enable multi-domain detection support */ modparam("domain", "db_url", "mysql://openser:openserrw@192.168.0.10 /openser") modparam("domain", "db_mode", 1) # Use caching # ----- multi-module params ----- /* uncomment the following line if you want to enable multi-domain support in the modules (dafault off) */ modparam("alias_db|auth_db|usrloc|uri_db", "use_domain", 1) # ----- presence params ----- /* uncomment the following lines if you want to enable presence */ #modparam("presence|presence_xml", "db_url", # "mysql://openser:openserrw@192.168.1.3/openser_1_3") #modparam("presence_xml", "force_active", 1) #modparam("presence", "server_address", "sip:192.168.1.2:5060") ####### Routing Logic ######## # main request routing logic route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(1); # do accouting ... setflag(3); # ... even if the transaction fails } route(1); } else { sl_send_reply("404","Not here"); } exit; } #initial requests # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); # DESCOMENTEI O BLOCO IF{} # authenticate if from local subscriber (uncomment to enable auth) if (!(method=="REGISTER") && from_uri==myself) { if (!proxy_authorize("", "subscriber")) { proxy_challenge("", "0"); exit; } if (!check_from()) { sl_send_reply("403","Forbidden auth ID"); exit; } consume_credentials(); # caller authenticated } # record routing if (!is_method("REGISTER|MESSAGE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(1); # do accouting } if (!uri==myself) /* replace with following line if multi-domain support is used */ ##if (!is_uri_host_local()) { append_hf("P-hint: outbound\r\n"); # if you have some interdomain connections via TLS ##if($rd=="tls_domain1.net") { ## t_relay("tls:domain1.net"); ## exit; ##} else if($rd=="tls_domain2.net") { ## t_relay("tls:domain2.net"); ## exit; ##} route(1); } # requests for my domain /* uncomment this if you want to enable presence server and comment the next 'if' block NOTE: uncomment also the definition of route[2] from below */ ##if( is_method("PUBLISH|SUBSCRIBE")) ## route(2); if (is_method("PUBLISH")) { sl_send_reply("503", "Service Unavailable"); exit; } if (is_method("REGISTER")) { # ACRESCENTEI O IP ADDR - MRG # authenticate the REGISTER requests (uncomment to enable auth) #if (!www_authorize("vmw02-srv", "subscriber")) #{ # www_challenge("vmw02-srv", "0"); # exit; #} # #if (!check_to()) ##{ ## sl_send_reply("403","Forbidden auth ID"); ## exit; ##} if (!save("location")) sl_reply_error(); exit; } if ($rU==NULL) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # apply DB based aliases (uncomment to enable) alias_db_lookup("dbaliases"); if (!lookup("location")) { switch ($retcode) { case -1: case -3: t_newtran(); t_reply("404", "Not Found"); exit; case -2: sl_send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also setflag(2); route(1); } route[1] { # for INVITEs enable some additional helper routes if (is_method("INVITE")) { t_on_branch("2"); t_on_reply("2"); t_on_failure("1"); } if (!t_relay()) { sl_reply_error(); }; exit; } # Presence route /* uncomment the whole following route for enabling presence NOTE: do not forget to enable the call of this route from the main route */ ##route[2] ##{ ## if (!t_newtran()) ## { ## sl_reply_error(); ## exit; ## }; ## ## if(is_method("PUBLISH")) ## { ## handle_publish(); ## t_release(); ## } ## else ## if( is_method("SUBSCRIBE")) ## { ## handle_subscribe(); ## t_release(); ## } ## ## exit; ##} branch_route[2] { xlog("new branch at $ru\n"); } onreply_route[2] { xlog("incoming reply\n"); } failure_route[1] { if (t_was_cancelled()) { exit; } # uncomment the following lines if you want to block client # redirect based on 3xx replies. ##if (t_check_status("3[0-9][0-9]")) { ##t_reply("404","Not found"); ## exit; ##} # uncomment the following lines if you want to redirect the failed # calls to a different new destination ##if (t_check_status("486|408")) { ## sethostport("192.168.2.100:5060"); ## append_branch(); ## # do not set the missed call flag again ## t_relay(); ##} } *Ngrep -d eth1 -qt -W byline port 5060* root@vmw02-srv:~# ngrep -d eth1 -qt -W byline port 5060 interface: eth1 (192.168.0.0/255.255.255.0) filter: (ip or ip6) and ( port 5060 ) U 2008/03/20 13:57:31.763060 192.168.0.25:5060 -> 192.168.0.10:5060 REGISTER sip:spo.voip.br SIP/2.0. Via: SIP/2.0/UDP 192.168.0.25;rport;branch=z9hG4bKignjupsg. Max-Forwards: 70. To: "vmwcli03" <sip:vmwcli03@spo.voip.br>. From: "vmwcli03" <sip:vmwcli03@spo.voip.br>;tag=dcoan. Call-ID: iswtwciijecswfp(a)192.168.0.25. CSeq: 883 REGISTER. Contact: <sip:vmwcli03@192.168.0.25>;expires=3600. Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO. User-Agent: Twinkle/1.0. Content-Length: 0. . U 2008/03/20 13:57:32.333159 192.168.0.25:5060 -> 192.168.0.10:5060 REGISTER sip:spo.voip.br SIP/2.0. Via: SIP/2.0/UDP 192.168.0.25;rport;branch=z9hG4bKignjupsg. Max-Forwards: 70. To: "vmwcli03" <sip:vmwcli03@spo.voip.br>. From: "vmwcli03" <sip:vmwcli03@spo.voip.br>;tag=dcoan. Call-ID: iswtwciijecswfp(a)192.168.0.25. CSeq: 883 REGISTER. Contact: <sip:vmwcli03@192.168.0.25>;expires=3600. Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO. User-Agent: Twinkle/1.0. Content-Length: 0. . U 2008/03/20 13:57:33.321952 192.168.0.25:5060 -> 192.168.0.10:5060 REGISTER sip:spo.voip.br SIP/2.0. Via: SIP/2.0/UDP 192.168.0.25;rport;branch=z9hG4bKignjupsg. Max-Forwards: 70. To: "vmwcli03" <sip:vmwcli03@spo.voip.br>. From: "vmwcli03" <sip:vmwcli03@spo.voip.br>;tag=dcoan. Call-ID: iswtwciijecswfp(a)192.168.0.25. CSeq: 883 REGISTER. Contact: <sip:vmwcli03@192.168.0.25>;expires=3600. Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO. User-Agent: Twinkle/1.0. Content-Length: 0. . *U 2008/03/20 13:57:35.309584 192.168.0.25:5060 -> 192.168.0.10:5060* *REGISTER sip:spo.voip.br SIP/2.0.* Via: SIP/2.0/UDP 192.168.0.25;rport;branch=z9hG4bKignjupsg. Max-Forwards: 70. To: "vmwcli03" <sip:vmwcli03@spo.voip.br>. From: "vmwcli03" <sip:vmwcli03@spo.voip.br>;tag=dcoan. Call-ID: iswtwciijecswfp(a)192.168.0.25. CSeq: 883 REGISTER. Contact: <sip:vmwcli03@192.168.0.25>;expires=3600. Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO. User-Agent: Twinkle/1.0. Content-Length: 0. . *U 2008/03/20 13:57:37.026703 192.168.0.10:5060 -> 192.168.0.25:5060* *SIP/2.0 483 Too Many Hops.* Via: SIP/2.0/UDP 192.168.0.25;rport=5060;branch=z9hG4bKignjupsg. To: "vmwcli03" <sip:vmwcli03@spo.voip.br>;tag= 1333af5435552d3677a0ca954c3a6e9f.6889. From: "vmwcli03" <sip:vmwcli03@spo.voip.br>;tag=dcoan. Call-ID: iswtwciijecswfp(a)192.168.0.25. CSeq: 883 REGISTER. Server: OpenSER (1.3.0-notls (i386/linux)). Content-Length: 0.

2 1

[OpenSER-Users] OpenSER Sending 483 Too Many Hops on register
by Marcio 25 Mar '08

25 Mar '08

Hi, If anybody can take a look openser.cfg below. I`m working vmware (enviroments virtual)...Whenever I try a "register" a client I get 483: Too many Hops....as if he had a loop (config file) but I am unable to find the problem... I use OpenSER 1.3-notls, client softphone: twincle or xlite... OpenSER - 192.168.0.10 Client - 192.168.0.25 Bind9/named.conf(SRV, A): spo.voip.br Does anyone know how to fix this ? Any hints.....?? thx... OpenSER File: ####### Global Parameters ######### alias="vmw02-srv.spo.voip.br" debug=6 log_stderror=no log_facility=LOG_LOCAL0 log_name="Openser" fork=yes children=8 tcp_children=4 group="openser" user="openser" /* uncomment the next line to disable TCP (default on) */ disable_tcp=yes /* uncomment the next line to enable the auto temporary blacklisting of not available destinations (default disabled) */ #disable_dns_blacklist=no /* uncomment the next line to enable IPv6 lookup after IPv4 dns lookup failures (default disabled) */ #dns_try_ipv6=yes /* uncomment the next line to disable the auto discovery of local aliases based on revers DNS on IPs (default on) */ #auto_aliases=no /* uncomment the following lines to enable TLS support (default off) */ #disable_tls = no #listen = tls:your_IP:5061 #tls_verify_server = 1 #tls_verify_client = 1 #tls_require_client_certificate = 0 #tls_method = TLSv1 #tls_certificate = "/usr/local/etc/openser/tls/user/user-cert.pem" #tls_private_key = "/usr/local/etc/openser/tls/user/user-privkey.pem" #tls_ca_list = "/usr/local/etc/openser/tls/user/user-calist.pem" port=5060 /* uncomment and configure the following line if you want openser to bind on a specific interface/port/proto (default bind on all available) */ listen=udp:192.168.0.10:5060 ####### Modules Section ######## #set module path mpath="/usr/local/lib/openser/modules/" /* uncomment next line for MySQL DB support */ loadmodule "mysql.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" modparam("maxfwd", "max_limit", 256) loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "mi_fifo.so" loadmodule "uri_db.so" loadmodule "uri.so" loadmodule "xlog.so" loadmodule "acc.so" loadmodule "avpops.so" /* uncomment next lines for MySQL based authentication support NOTE: a DB (like mysql) module must be also loaded */ loadmodule "auth.so" loadmodule "auth_db.so" /* uncomment next line for aliases support NOTE: a DB (like mysql) module must be also loaded */ loadmodule "alias_db.so" /* uncomment next line for multi-domain support NOTE: a DB (like mysql) module must be also loaded NOTE: be sure and enable multi-domain support in all used modules (see "multi-module params" section ) */ loadmodule "domain.so" /* uncomment the next two lines for presence server support NOTE: a DB (like mysql) module must be also loaded */ #loadmodule "presence.so" #loadmodule "presence_xml.so" # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) #modparam("rr", "append_fromtag", 0) # ----- rr params ----- #modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam("registrar", "max_contacts", 10) # ----- uri_db params ----- /* by default we disable the DB support in the module as we do not need it in this configuration */ modparam("uri_db", "use_uri_table", 0) modparam("uri_db", "db_url", "") # ----- acc params ----- /* what sepcial events should be accounted ? */ modparam("acc", "early_media", 1) modparam("acc", "report_ack", 1) modparam("acc", "report_cancels", 1) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "failed_transaction_flag", 3) modparam("acc", "log_flag", 1) modparam("acc", "log_missed_flag", 2) /* uncomment the following lines to enable DB accounting also */ modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 2) # ----- usrloc params ----- #modparam("usrloc", "db_mode", 0) /* uncomment the following lines if you want to enable DB persistency for location entries */ modparam("usrloc", "db_mode", 2) modparam("usrloc", "db_url", "mysql://openser:openserrw@192.168.0.10 /openser") modparam("usrloc", "user_column", "username") modparam("usrloc", "domain_column", "domain") modparam("usrloc", "contact_column", "contact") modparam("usrloc", "expires_column", "expires") modparam("usrloc", "q_column", "q") modparam("usrloc", "callid_column", "callid") modparam("usrloc", "cseq_column", "cseq") modparam("usrloc", "methods_column", "methods") modparam("usrloc", "flags_column", "flags") modparam("usrloc", "user_agent_column", "user_agent") modparam("usrloc", "received_column", "received") modparam("usrloc", "socket_column", "socket") modparam("usrloc", "use_domain", 0) modparam("usrloc", "desc_time_order", 0) modparam("usrloc", "timer_interval", 60) modparam("usrloc", "matching_mode", 0) modparam("usrloc", "cseq_delay", 20) # ----- auth_db params ----- /* uncomment the following lines if you want to enable the DB based authentication */ #modparam("auth_db", "calculate_ha1", yes) #modparam("auth_db", "password_column", "password") modparam("auth_db", "db_url", "mysql://openser:openserrw@192.168.0.10 /openser") #modparam("auth_db", "load_credentials", "") # ----- alias_db params ----- /* uncomment the following lines if you want to enable the DB based aliases */ modparam("alias_db", "db_url", "mysql://openser:openserrw@192.168.0.10 /openser") # ----- domain params ----- /* uncomment the following lines to enable multi-domain detection support */ modparam("domain", "db_url", "mysql://openser:openserrw@192.168.0.10 /openser") modparam("domain", "db_mode", 1) # Use caching # ----- multi-module params ----- /* uncomment the following line if you want to enable multi-domain support in the modules (dafault off) */ modparam("alias_db|auth_db|usrloc|uri_db", "use_domain", 1) # ----- presence params ----- /* uncomment the following lines if you want to enable presence */ #modparam("presence|presence_xml", "db_url", # "mysql://openser:openserrw@192.168.1.3/openser_1_3") #modparam("presence_xml", "force_active", 1) #modparam("presence", "server_address", "sip:192.168.1.2:5060") ####### Routing Logic ######## # main request routing logic route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(1); # do accouting ... setflag(3); # ... even if the transaction fails } route(1); } else { sl_send_reply("404","Not here"); } exit; } #initial requests # CANCEL processing if (is_method("CANCEL")){ if (t_check_trans()) t_relay(); exit; } t_check_trans(); # authenticate if from local subscriber (uncomment to enable auth) if (!(method=="REGISTER") && from_uri==myself) { if (!proxy_authorize("", "subscriber")) { proxy_challenge("", "0"); exit; } if (!check_from()) { sl_send_reply("403","Forbidden auth ID"); exit; } consume_credentials(); # caller authenticated } # record routing if (!is_method("REGISTER|MESSAGE")) { record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(1); # do accouting } if (!uri==myself) /* replace with following line if multi-domain support is used */ ##if (!is_uri_host_local()) { append_hf("P-hint: outbound\r\n"); # if you have some interdomain connections via TLS ##if($rd=="tls_domain1.net") { ## t_relay("tls:domain1.net"); ## exit; ##} else if($rd=="tls_domain2.net") { ## t_relay("tls:domain2.net"); ## exit; ##} route(1); } # requests for my domain /* uncomment this if you want to enable presence server and comment the next 'if' block NOTE: uncomment also the definition of route[2] from below */ ##if( is_method("PUBLISH|SUBSCRIBE")) ## route(2); if (is_method("PUBLISH")){ sl_send_reply("503", "Service Unavailable"); exit; } if (is_method("REGISTER")){ # ACRESCENTEI O IP ADDR - MRG # authenticate the REGISTER requests (uncomment to enable auth) #if (!www_authorize("vmw02-srv", "subscriber")){ # www_challenge("vmw02-srv", "0"); # exit; #} # #if (!check_to()){ ## sl_send_reply("403","Forbidden auth ID"); ## exit; ##} if (!save("location")) sl_reply_error(); exit; } if ($rU==NULL) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # apply DB based aliases (uncomment to enable) alias_db_lookup("dbaliases"); if (!lookup("location")) { switch ($retcode) { case -1: case -3: t_newtran(); t_reply("404", "Not Found"); exit; case -2: sl_send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also setflag(2); route(1); } route[1] { # for INVITEs enable some additional helper routes if (is_method("INVITE")) { t_on_branch("2"); t_on_reply("2"); t_on_failure("1"); } if (!t_relay()) { sl_reply_error(); }; exit; } # Presence route /* uncomment the whole following route for enabling presence NOTE: do not forget to enable the call of this route from the main route */ ##route[2] ##{ ## if (!t_newtran()) ## { ## sl_reply_error(); ## exit; ## }; ## ## if(is_method("PUBLISH")) ## { ## handle_publish(); ## t_release(); ## } ## else ## if( is_method("SUBSCRIBE")) ## { ## handle_subscribe(); ## t_release(); ## } ## ## exit; ##} branch_route[2] { xlog("new branch at $ru\n"); } onreply_route[2] { xlog("incoming reply\n"); } failure_route[1] { if (t_was_cancelled()) { exit; } # uncomment the following lines if you want to block client # redirect based on 3xx replies. ##if (t_check_status("3[0-9][0-9]")) { ##t_reply("404","Not found"); ## exit; ##} # uncomment the following lines if you want to redirect the failed # calls to a different new destination ##if (t_check_status("486|408")) { ## sethostport("192.168.2.100:5060"); ## append_branch(); ## # do not set the missed call flag again ## t_relay(); ##} } Ngrep -d eth1 -qt -W byline port 5060 root@vmw02-srv:~# ngrep -d eth1 -qt -W byline port 5060 interface: eth1 (192.168.0.0/255.255.255.0) filter: (ip or ip6) and ( port 5060 ) U 2008/03/20 13:57:31.763060 192.168.0.25:5060 -> 192.168.0.10:5060 REGISTER sip:spo.voip.br SIP/2.0. Via: SIP/2.0/UDP 192.168.0.25;rport;branch=z9hG4bKignjupsg. Max-Forwards: 70. To: "vmwcli03" <sip:vmwcli03@spo.voip.br>. From: "vmwcli03" <sip:vmwcli03@spo.voip.br>;tag=dcoan. Call-ID: iswtwciijecswfp(a)192.168.0.25. CSeq: 883 REGISTER. Contact: <sip:vmwcli03@192.168.0.25>;expires=3600. Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO. User-Agent: Twinkle/1.0. Content-Length: 0. . U 2008/03/20 13:57:32.333159 192.168.0.25:5060 -> 192.168.0.10:5060 REGISTER sip:spo.voip.br SIP/2.0. Via: SIP/2.0/UDP 192.168.0.25;rport;branch=z9hG4bKignjupsg. Max-Forwards: 70. To: "vmwcli03" <sip:vmwcli03@spo.voip.br>. From: "vmwcli03" <sip:vmwcli03@spo.voip.br>;tag=dcoan. Call-ID: iswtwciijecswfp(a)192.168.0.25. CSeq: 883 REGISTER. Contact: <sip:vmwcli03@192.168.0.25>;expires=3600. Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO. User-Agent: Twinkle/1.0. Content-Length: 0. . U 2008/03/20 13:57:33.321952 192.168.0.25:5060 -> 192.168.0.10:5060 REGISTER sip:spo.voip.br SIP/2.0. Via: SIP/2.0/UDP 192.168.0.25;rport;branch=z9hG4bKignjupsg. Max-Forwards: 70. To: "vmwcli03" <sip:vmwcli03@spo.voip.br>. From: "vmwcli03" <sip:vmwcli03@spo.voip.br>;tag=dcoan. Call-ID: iswtwciijecswfp(a)192.168.0.25. CSeq: 883 REGISTER. Contact: <sip:vmwcli03@192.168.0.25>;expires=3600. Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO. User-Agent: Twinkle/1.0. Content-Length: 0. . U 2008/03/20 13:57:35.309584 192.168.0.25:5060 -> 192.168.0.10:5060 REGISTER sip:spo.voip.br SIP/2.0. Via: SIP/2.0/UDP 192.168.0.25;rport;branch=z9hG4bKignjupsg. Max-Forwards: 70. To: "vmwcli03" <sip:vmwcli03@spo.voip.br>. From: "vmwcli03" <sip:vmwcli03@spo.voip.br>;tag=dcoan. Call-ID: iswtwciijecswfp(a)192.168.0.25. CSeq: 883 REGISTER. Contact: <sip:vmwcli03@192.168.0.25>;expires=3600. Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO. User-Agent: Twinkle/1.0. Content-Length: 0. . U 2008/03/20 13:57:37.026703 192.168.0.10:5060 -> 192.168.0.25:5060 SIP/2.0 483 Too Many Hops. Via: SIP/2.0/UDP 192.168.0.25;rport=5060;branch=z9hG4bKignjupsg. To: "vmwcli03" <sip:vmwcli03@spo.voip.br>;tag= 1333af5435552d3677a0ca954c3a6e9f.6889. From: "vmwcli03" <sip:vmwcli03@spo.voip.br>;tag=dcoan. Call-ID: iswtwciijecswfp(a)192.168.0.25. CSeq: 883 REGISTER. Server: OpenSER (1.3.0-notls (i386/linux)). Content-Length: 0.

4 4

[OpenSER-Users] Info about redirect function
by Valerio Di Marino 25 Mar '08

25 Mar '08

Good morning to all. I am Valerio, and this im my first post. I was never using OPENSER: I discovered it this morning, searcing for a solution. I have this situation: I am using Motorola Canopy wireless antennas to deploy internet connection, but it implements a symmetric NAT which make not operating STUN protocol. I think that one of solution should be to install a proxy server which redirects registering and calls to the public proxy (euteliavoip). I thought that OPENSER should be a good solution: can you confirm this? Thanks for every your suggestions. Valerio

2 1

[OpenSER-Users] Openser fails to load module snmpstats
by Stanislaw Pitucha 25 Mar '08

25 Mar '08

Reposting - for some reason original message got attached as response to another thread on list, so some people might have missed it. --- Hey I can't get snmpstats module to load. Message from logs is (during config parsing): ERROR:core:sr_load_module: could not open module </usr/local/openser/lib/openser/modules/snmpstats.so>: /usr/lib/libnetsnmpagent.so.10: undefined symbol: boot_DynaLoader System: CentOS-5.1, 32b, up-to-date. Openser: svn head from 1.3 branch Tried `ldconfig`, recompiling after `make clean`, ... Has anyone got some ideas how to resolve it? Google points to problems with linking C extensions for perl only, which I think is not the case here. Thanks, Stanisław Pitucha

1 0

[OpenSER-Users] Openser sending 513 Message Too Big
by Aadilkhan Maniyar 25 Mar '08

25 Mar '08

Hi, Please find my openser.cfg file below. Whenever I try to call a softphone registered with Openser from another softphone (also registered with Openser) i get a 513 Message Too Big. I guess there is a loop in the configuration file but I am unable to find it out. Any help would be appreciated. Regards, Aadil ------------------------------------------------------------------------ ------------------------------------------------------------------------ ---------- ####### Global Parameters ######### debug=9 log_stderror=no log_facility=LOG_LOCAL0 fork=yes children=4 /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to enable the auto temporary blacklisting of not available destinations (default disabled) */ #disable_dns_blacklist=no /* uncomment the next line to enable IPv6 lookup after IPv4 dns lookup failures (default disabled) */ #dns_try_ipv6=yes /* uncomment the next line to disable the auto discovery of local aliases based on revers DNS on IPs (default on) */ #auto_aliases=no /* uncomment the following lines to enable TLS support (default off) */ #disable_tls = no #listen = tls:your_IP:5061 #tls_verify_server = 1 #tls_verify_client = 1 #tls_require_client_certificate = 0 #tls_method = TLSv1 #tls_certificate = "/opt/openser/etc/openser/tls/user/user-cert.pem" #tls_private_key = "/opt/openser/etc/openser/tls/user/user-privkey.pem" #tls_ca_list = "/opt/openser/etc/openser/tls/user/user-calist.pem" port=5060 /* uncomment and configure the following line if you want openser to bind on a specific interface/port/proto (default bind on all available) */ listen=udp:IP1:5060 listen=udp:IP2:5060 ####### Modules Section ######## #set module path mpath="/opt/openser/lib/openser/modules/" /* uncomment next line for MySQL DB support */ 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 "mi_fifo.so" loadmodule "uri_db.so" loadmodule "uri.so" loadmodule "xlog.so" loadmodule "acc.so" loadmodule "avpops.so" /* uncomment next lines for MySQL based authentication support NOTE: a DB (like mysql) module must be also loaded */ loadmodule "auth.so" loadmodule "auth_db.so" /* uncomment next line for aliases support NOTE: a DB (like mysql) module must be also loaded */ #loadmodule "alias_db.so" /* uncomment next line for multi-domain support NOTE: a DB (like mysql) module must be also loaded NOTE: be sure and enable multi-domain support in all used modules (see "multi-module params" section ) */ loadmodule "domain.so" /* uncomment the next two lines for presence server support NOTE: a DB (like mysql) module must be also loaded */ #loadmodule "presence.so" # ----------------- setting module-specific parameters --------------- modparam("usrloc|auth_db|avpops|domain", "db_url", "mysql://openser:openserrw@localhost/openser") # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- rr params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam("registrar", "max_contacts", 10) # ----- uri_db params ----- /* by default we disable the DB support in the module as we do not need it in this configuration */ modparam("uri_db", "use_uri_table", 0) modparam("uri_db", "db_url", "") # ----- acc params ----- /* what sepcial events should be accounted ? */ modparam("acc", "early_media", 1) modparam("acc", "report_ack", 1) modparam("acc", "report_cancels", 1) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "failed_transaction_flag", 3) modparam("acc", "log_flag", 1) modparam("acc", "log_missed_flag", 2) /* uncomment the following lines to enable DB accounting also */ modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 2) # ----- usrloc params ----- #modparam("usrloc", "db_mode", 0) /* uncomment the following lines if you want to enable DB persistency for location entries */ modparam("usrloc", "db_mode", 1) # ----- auth_db params ----- /* uncomment the following lines if you want to enable the DB based authentication */ modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") # ----- alias_db params ----- /* uncomment the following lines if you want to enable the DB based aliases */ #modparam("alias_db", "db_url", # "mysql://openser:openserrw@server/openser") # ----- domain params ----- /* uncomment the following lines to enable multi-domain detection support */ modparam("domain", "db_mode", 1) # Use caching # ----- multi-module params ----- /* uncomment the following line if you want to enable multi-domain support in the modules (dafault off) */ #modparam("alias_db|auth_db|usrloc|uri_db", "use_domain", 1) # ----- presence params ----- /* uncomment the following lines if you want to enable presence */ #modparam("presence", "db_url", # "mysql://openser:openserrw@server/openser") #modparam("presence", "force_active", 1) #modparam("presence", "server_address", "sip:192.168.1.2:5060") modparam("avpops", "avp_table", "usr_preferences") ####### Routing Logic ######## # main request 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"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; # 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); }; if (!uri==myself) { if (method=="REGISTER") { # if (!www_authorize("", "subscriber")) { #www_challenge("", "0"); #exit; #} # #save("location"); #exit; #xlog("L_DEBUG", "Unknown Domain"); #sl_send_reply("403", "Forbidden"); }; if (method=="INVITE") { xlog("L_DEBUG", "INVITE for other domain"); route(4); }; # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); }; # 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") { if (!www_authorize("", "subscriber")) { www_challenge("", "0"); exit; }; save("location"); exit; }; # requests for Media server #if(is_method("INVITE") && !has_totag() ) { if(is_method("INVITE")) { ;route(3); exit; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { if(isflagset(2)) { # route to Asterisk Media Server prefix("1"); rewritehostport("asterisk:6060"); route(1); } else { sl_send_reply("404", "Not Found"); exit; } }; append_hf("P-hint: usrloc applied\r\n"); }; route(1); } route[1] { xlog("L_INFO","Entered route 1\n"); if(isflagset(2)) t_on_failure("1"); if (!t_relay()) { sl_reply_error(); }; exit; } route[3] { xlog("L_INFO","Entered route 3\n"); t_on_reply("3"); # Authenticate calls #if(!proxy_authorize("","subscriber")) { # proxy_challenge("","0"); # exit; #} # end Authenticate calls # route to Asterisk Media Server #rewritehostport("asterisk:6060"); route(1); } #Route for calls to other domains route[4] { xlog("L_INFO","Entered route 4"); t_on_reply("4"); rewritehostport("asterisk:6060"); if (!t_relay()) { sl_reply_error(); }; exit; } failure_route[1] { if (t_was_cancelled()) { xdbg("transaction was cancelled by UAC\n"); return; } # restore initial uri avp_pushto("$ruri","$avp(i:10)"); prefix("1"); # route to Asterisk Media Server rewritehostport("asterisk:6060"); resetflag(2); route(1); } failure_route[4] { t_relay(); } onreply_route[3] { if(status =~"180") { t_on_failure("4"); }; } ------------------------------------------------------------------------ ------------------------------------------------------------------------ ----------

2 1

[OpenSER-Users] OpenSER location table timezones
by Adrian A 25 Mar '08

25 Mar '08

I have an interesting problem with an OpenSER setup with 2 proxies in different timezones but using a shared DB with usrloc db_mode 2 (it actually is a master-master replicated MySQL but I don't think that is relevant). The issue is that when clients register to the proxy located in the PST zone, the 'expires' time is much earlier than the current time in EST. As a result, the proxy in EST deletes the entry from the location table since it thinks it is stale. Is there any way around this, other than setting both proxy servers to the same timezone?

3 4

[Serusers] book of asterisk with ser
by aka 25 Mar '08

25 Mar '08

dear sir i want your asterisk and ser ecopy.please send me details about it

1 0

Re: [OpenSER-Users] LCR Gateway Priority....
by Douglas Garstang 25 Mar '08

25 Mar '08

>Douglas Garstang wrote: >> >Douglas Garstang writes: >> >> The LCR module actually works better. It the gatways had a priority, >> we'd be happy! >> > >You have the source, go add support for priorities... > > >Should be as simple as tweaking the SQL statement ----> 'ORDER BY priority' > Actually, no. I don't think so. If OpenSER picks a gateway randomly, it probably does this in memory. Returning gateways in a specific order will have no effect if OpenSER picks one at random from an in-memory list. Doug. ____________________________________________________________________________________ Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs

1 0

Re: [OpenSER-Users] Carrierroute
by Douglas Garstang 25 Mar '08

25 Mar '08

After spending all day working with carrierroute, I think I am missing something pretty fundamental about it. I see that I can use cr_tree_rewrite_uri() to select a new carrier, but WHY does the carrier have to be hard coded in the cfg file, and why does it require a STRING name of the carrier? I really don't understand that. It makes no sense whatsoever to hard code your carriers in your config file. Doug. ----- Original Message ---- From: Douglas Garstang <dougmig33(a)yahoo.com> To: Ovidiu Sas <osas(a)voipembedded.com> Cc: users(a)lists.openser.org Sent: Monday, March 24, 2008 7:59:13 PM Subject: Re: [OpenSER-Users] Carrierroute Ovidiu, Ok... thanks... now... do you know how I can change carrier mid call? Doug. ----- Original Message ---- From: Ovidiu Sas <osas(a)voipembedded.com> To: Douglas Garstang <dougmig33(a)yahoo.com> Cc: users(a)lists.openser.org Sent: Monday, March 24, 2008 7:24:31 PM Subject: Re: [OpenSER-Users] Carrierroute The database config is right in the README file. Here are the trees: +----+----------+ | id | carrier | +----+----------+ | 1 | carrier1 | | 2 | carrier2 | | 3 | default | +----+----------+ I agree with you that the names are confusing :) You may or may not need to use the subscriber tables, based on what functions are you using. In the example that I gave you, you don't need the location table. Instead, you force a routing tree (in my example is the "default" tree). Hope this helps, Ovidiu Sas On Mon, Mar 24, 2008 at 9:51 PM, Douglas Garstang <dougmig33(a)yahoo.com> wrote: > > Ovidiu, > > I need database config info.... your example uses the config file. > What is a 'tree'? The documentation keeps talking about it, but I don't see > it defined anywhere. Carrierroute SEEMS to need the subscriber table, but it > doesn't load it from the database.... why not? > > Why should I spend days experimenting to get it to work? If the docs where > adequate, that wouldn't be required. > > Doug. > > > > ----- Original Message ---- > From: Ovidiu Sas <osas(a)voipembedded.com> > To: Douglas Garstang <dougmig33(a)yahoo.com> > Cc: users(a)lists.openser.org > Sent: Monday, March 24, 2008 5:43:53 PM > Subject: Re: [OpenSER-Users] Carrierroute > > Hello Douglas, > > The database sample is available in the README file here: > http://www.openser.org/docs/modules/1.3.x/carrierroute.html#AEN575 > The entire documentation is there, you just need to read it carefully > and experiment. > > You had a similar post about the documentation for the lcr module and > in the end it seems that you figure it out :) > If you think that you can improve documentation, feel free to send > patches for the README file. > > Here you can find a configuration file example for carrierroute: > http://voipembedded.com/resources/openser_cr.cfg > > > Regards, > Ovidiu Sas > > > On Mon, Mar 24, 2008 at 7:52 PM, Douglas Garstang <dougmig33(a)yahoo.com> > wrote: > > > > I have been trying to get the carrierroute module to work ALL day. > > Can someone 'supplement' the openser docs with maybe some sample database > > data? > > > > The docs really aren't making sense. The docs keep talking about a routing > > tree? What is a routing tree? > > When I start openser, I don't see it query the subscriber table. Should > it? > > > > Thanks, > > Doug. > > > > > > __________________________________________________ > > Do You Yahoo!? > > Tired of spam? Yahoo! Mail has the best spam protection around > > http://mail.yahoo.com > > _______________________________________________ > > Users mailing list > > Users(a)lists.openser.org > > http://lists.openser.org/cgi-bin/mailman/listinfo/users > > > > > > > ________________________________ > Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it > now. Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. -----Inline Attachment Follows----- _______________________________________________ Users mailing list Users(a)lists.openser.org http://lists.openser.org/cgi-bin/mailman/listinfo/users ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ

1 0

[Serusers] Problems running ser 0.9.7
by Ricardo Martinez 25 Mar '08

25 Mar '08

Hello List. I'm having a problema trying to run ser-0.9.7. The compilation was fine.. but i'm not able to start SER due to this error : Mar 25 00:42:53 edgelcr ser: loading module /usr/local/lib/ser/modules/acc.so Mar 25 00:42:53 edgelcr ser: 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 Mar 25 00:42:53 edgelcr ser: parse error (88,13-14): failed to load module Can someone help me here? Thanks in advance Regards, Ricardo Martinez.-

1 0

Re: [OpenSER-Users] Carrierroute
by Douglas Garstang 25 Mar '08

25 Mar '08

Ovidiu, Ok... thanks... now... do you know how I can change carrier mid call? Doug. ----- Original Message ---- From: Ovidiu Sas <osas(a)voipembedded.com> To: Douglas Garstang <dougmig33(a)yahoo.com> Cc: users(a)lists.openser.org Sent: Monday, March 24, 2008 7:24:31 PM Subject: Re: [OpenSER-Users] Carrierroute The database config is right in the README file. Here are the trees: +----+----------+ | id | carrier | +----+----------+ | 1 | carrier1 | | 2 | carrier2 | | 3 | default | +----+----------+ I agree with you that the names are confusing :) You may or may not need to use the subscriber tables, based on what functions are you using. In the example that I gave you, you don't need the location table. Instead, you force a routing tree (in my example is the "default" tree). Hope this helps, Ovidiu Sas On Mon, Mar 24, 2008 at 9:51 PM, Douglas Garstang <dougmig33(a)yahoo.com> wrote: > > Ovidiu, > > I need database config info.... your example uses the config file. > What is a 'tree'? The documentation keeps talking about it, but I don't see > it defined anywhere. Carrierroute SEEMS to need the subscriber table, but it > doesn't load it from the database.... why not? > > Why should I spend days experimenting to get it to work? If the docs where > adequate, that wouldn't be required. > > Doug. > > > > ----- Original Message ---- > From: Ovidiu Sas <osas(a)voipembedded.com> > To: Douglas Garstang <dougmig33(a)yahoo.com> > Cc: users(a)lists.openser.org > Sent: Monday, March 24, 2008 5:43:53 PM > Subject: Re: [OpenSER-Users] Carrierroute > > Hello Douglas, > > The database sample is available in the README file here: > http://www.openser.org/docs/modules/1.3.x/carrierroute.html#AEN575 > The entire documentation is there, you just need to read it carefully > and experiment. > > You had a similar post about the documentation for the lcr module and > in the end it seems that you figure it out :) > If you think that you can improve documentation, feel free to send > patches for the README file. > > Here you can find a configuration file example for carrierroute: > http://voipembedded.com/resources/openser_cr.cfg > > > Regards, > Ovidiu Sas > > > On Mon, Mar 24, 2008 at 7:52 PM, Douglas Garstang <dougmig33(a)yahoo.com> > wrote: > > > > I have been trying to get the carrierroute module to work ALL day. > > Can someone 'supplement' the openser docs with maybe some sample database > > data? > > > > The docs really aren't making sense. The docs keep talking about a routing > > tree? What is a routing tree? > > When I start openser, I don't see it query the subscriber table. Should > it? > > > > Thanks, > > Doug. > > > > > > __________________________________________________ > > Do You Yahoo!? > > Tired of spam? Yahoo! Mail has the best spam protection around > > http://mail.yahoo.com > > _______________________________________________ > > Users mailing list > > Users(a)lists.openser.org > > http://lists.openser.org/cgi-bin/mailman/listinfo/users > > > > > > > ________________________________ > Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it > now. ____________________________________________________________________________________ Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping

1 0

Re: [OpenSER-Users] LCR Gateway Priority....
by Douglas Garstang 25 Mar '08

25 Mar '08

1.3.1... ----- Original Message ---- From: Ovidiu Sas <osas(a)voipembedded.com> To: Jeremy McNamara <jj(a)nufone.net> Cc: Douglas Garstang <dougmig33(a)yahoo.com>; users(a)lists.openser.org Sent: Monday, March 24, 2008 5:34:30 PM Subject: Re: [OpenSER-Users] LCR Gateway Priority.... On Mon, Mar 24, 2008 at 7:27 PM, Jeremy McNamara <jj(a)nufone.net> wrote: > Douglas Garstang wrote: > > > >Douglas Garstang writes: > > > > The LCR module actually works better. It the gatways had a priority, > > we'd be happy! > > You have the source, go add support for priorities... > > Should be as simple as tweaking the SQL statement ----> 'ORDER BY priority' Which version of openser are you using? The SQL statements are gone since 1.2 afair. Regards, Ovidiu Sas __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

2 1

[OpenSER-Users] Carrierroute
by Douglas Garstang 25 Mar '08

25 Mar '08

I have been trying to get the carrierroute module to work ALL day. Can someone 'supplement' the openser docs with maybe some sample database data? The docs really aren't making sense. The docs keep talking about a routing tree? What is a routing tree? When I start openser, I don't see it query the subscriber table. Should it? Thanks, Doug. __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

2 1

Re: [OpenSER-Users] Carrierroute
by Douglas Garstang 25 Mar '08

25 Mar '08

Actually, further to that, how can I switch to another carrier in route() ? I can't see any way in the docs to do this. ----- Original Message ---- From: Douglas Garstang <dougmig33(a)yahoo.com> To: users(a)lists.openser.org Sent: Monday, March 24, 2008 4:52:55 PM Subject: [OpenSER-Users] Carrierroute I have been trying to get the carrierroute module to work ALL day. Can someone 'supplement' the openser docs with maybe some sample database data? The docs really aren't making sense. The docs keep talking about a routing tree? What is a routing tree? When I start openser, I don't see it query the subscriber table. Should it? Thanks, Doug. __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com -----Inline Attachment Follows----- _______________________________________________ Users mailing list Users(a)lists.openser.org http://lists.openser.org/cgi-bin/mailman/listinfo/users __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

1 0

[Serusers] Help with checking header fields
by Steven C. Blair 24 Mar '08

24 Mar '08

I need help constructing an if statement to check for a specific header field. If my proxy receives an INVITE I would like to check further into the message body for a connection address with a value of all zeros (0.0.0.0). Does anyone on this list know the syntax for performing this check? Thanks,Steve

3 3

[Serusers] pls. help me on my installation
by Allan Jardinel 24 Mar '08

24 Mar '08

I'm using ubuntu 7.10 and install ser 0.9.7 and rtp, after the installation i test and it can call both users, when i try to restart the computer xten can not register anymore so i check the settings and its ok. but when i check the /var/run/ser.pid there is no file, also the /tmp/ser_fifo has no file so restart the ser using the /etc/ser restart or force-reload but still pid and fifo has no file. pls. help me to solve this problem.... Thank you.. Allan Send instant messages to your online friends http://uk.messenger.yahoo.com

2 1

[OpenSER-Users] LCR Gateway Priority....
by Douglas Garstang 24 Mar '08

24 Mar '08

Can anyone tell me why gateways in LCR don't have a priority? We have multiple POP's where we originate calls from. The ITSP's that we send these calls to ALSO have multiple POP's. Choosing the best gateway isn't just a matter of choosing the least cost... It's also a matter of choosing the closest POP for that ITSP. Being able to set a priority for each gateway means I could do this... mysql> select * from lcr; +----+--------+---------------+--------+----------+ | id | prefix | from_uri | grp_id | priority | +----+--------+---------------+--------+----------+ | 8 | 1303 | @203.84.213.8 | 1 | 1 | <-- (Hong Kong) | 10 | 1303 | @67.151.19.114| 2 | 1 | <-- (USA) +----+--------+---------------+--------+----------+ mysql> select * from gw; +----+---------------------+--------+------------+------+------------+-----------+-------+--------+----+ | id | gw_name | grp_id | ip_addr | port | uri_scheme | transport | strip | prefix | dm | Priority +----+---------------------+--------+------------+------+------------+-----------+-------+--------+----+ | 12 | TELEGLOBE_CH_HKG_01 | 1 | 1079401702 | 5060 | 1 | 1 | NULL | NULL | 1 | 1 | 13 | VERIZON_CA_YTO_01 | 1 | 3517395939 | 5060 | 1 | 1 | NULL | NULL | 1 | 2 | 14 | ARBINET_CH_HKG_01 | 1 | 3423099979 | 5060 | 1 | 1 | NULL | NULL | 1 | 3 | 15 | TELEGLOBE_US_LAX_01 | 2 | 3517395939 | 5060 | 1 | 1 | NULL | NULL | 1 | 1 | 16 | VERIZON_CA_YTO_02 | 2 | 1079401702 | 5060 | 1 | 1 | NULL | NULL | 1 | 2 | 17 | ARBINET_US_LGA_01 | 2 | 3423099979 | 5060 | 1 | 1 | NULL | NULL | 1 | 3 +----+---------------------+--------+------------+------+------------+-----------+-------+--------+----+ As you can see here, for calls to to prefix 1303 from 203.84.213.8(Hong Kong), we'd send calls to gateways TELEGLOBE_CH_HKG_01, VERIZON_CA_YTO_01 and ARBINET_CH_HKG_01 in that order. For calls to 1303 from 67.151.19.114(the US), we'd send calls to the same our ITSP's in the same order, but use gateways in different POPS, TELEGLOBE_US_LAX_01, VERIZON_CA_YTO_02, ARBINET_US_LGA_01. What's really important here is that the choice of gateway for the cheapest ITSP can be modified to suit the location. Now, WITHOUT the priority, I have to do THIS instead... mysql> select * from lcr; +----+--------+----------------+--------+----------+ | id | prefix | from_uri | grp_id | priority | +----+--------+----------------+--------+----------+ | 1 | 1303 | @203.84.213.8 | 1 | 1 | | 2 | 1303 | @203.84.213.8 | 2 | 2 | | 3 | 1303 | @203.84.213.8 | 3 | 3 | | 5 | 1303 | @67.151.19.114 | 4 | 1 | | 6 | 1303 | @67.151.19.114 | 5 | 2 | | 7 | 1303 | @67.151.19.114 | 6 | 3 | +----+--------+----------------+--------+----------+ mysql> select * from gw; +----+---------------------+--------+------------+------+------------+-----------+-------+--------+----+ | id | gw_name | grp_id | ip_addr | port | uri_scheme | transport | strip | prefix | dm | +----+---------------------+--------+------------+------+------------+-----------+-------+--------+----+ | 12 | TELEGLOBE_CH_HKG_01 | 1 | 1079401702 | 5060 | 1 | 1 | NULL | NULL | 1 | | 13 | VERIZON_CA_YTO_01 | 2 | 3517395939 | 5060 | 1 | 1 | NULL | NULL | 1 | | 14 | ARBINET_CH_HKG_01 | 3 | 3423099979 | 5060 | 1 | 1 | NULL | NULL | 1 | | 15 | TELEGLOBE_US_LAX_01 | 4 | 3517395939 | 5060 | 1 | 1 | NULL | NULL | 1 | | 16 | VERIZON_CA_YTO_02 | 5 | 1079401702 | 5060 | 1 | 1 | NULL | NULL | 1 | | 17 | ARBINET_US_LGA_01 | 6 | 3423099979 | 5060 | 1 | 1 | NULL | NULL | 1 | +----+---------------------+--------+------------+------+------------+-----------+-------+--------+----+ As you can see, solution 1 requires LESS columns in the database. What other options do I have? I guess I could create separate lcr and gw tables for every POP we have... yuck. Unfortunately, there is no priority column in the gateway table. :( I see someone else asked about it a while ago... http://www.mail-archive.com/devel@openser.org/msg06757.html Doug. ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ

2 1

[OpenSER-Users] LCR Failover
by Douglas Garstang 23 Mar '08

23 Mar '08

I'd like to think I'm not stupid, but I've been dealing with OpenSER for 2-3 years now and it still frustrates me just as much as it did the very first time I looked at it. Can LCR do failover? I have this below. I can see that one of my gateways is selected, seemingly at random, and then when it doesn't reply (because I used a bogus IP), it doesn't time-out after 3s and go to the next gw. Is this possible? How do I do it? Btw... can anyone recommend a book on OpenSER, since the documentation is so lacking? modparam("tm", "fr_inv_timer_avp", "$avp(i:705)") route{ $avp(i:705) = "3"; ... if (!lookup("location")) { switch ($retcode) { case -1: case -3: t_newtran(); sl_send_reply("100","Trying"); if (!load_gws()) { sl_send_reply("500", "Internal server error, unable to load gateways"); exit; } if(!next_gw()){ sl_send_reply("503", "Service not available, no gateways found"); exit; }; route(1); } } route[1] { if (is_method("INVITE")) { t_on_branch("2"); t_on_reply("2"); t_on_failure("1"); } if (!t_relay()) { sl_reply_error(); }; exit; } ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ

2 1

[Serusers] How do you force SDP c= ipaddr to be a specific value that isn't either rtpproxy interface IP?
by Frank Durda IV 23 Mar '08

23 Mar '08

I have a problem with SER 2.0.0-RC1 and rtpproxy that needs some help. I have a PSTN gateway config and a SER box all in a 10.x.x.x address space. Directly connected to two interfaces on the connect to a Cisco router, which performs 1:1 NAT conversion to provide public Internet IPs to those interfaces, one for SIP and one for RTP. (They are separate for bandwidth, debugging and access-list reasons.) Another two interfaces on the SER connect to the PSTN switch, which has separate ports for SIP and RTP traffic. Across the Internet or direct connections to the Cisco router are from Asterisk boxes or similar devices that proxy all SIP/RTP data from their SIP phones. It looks like this: Asterisk SIP SIP SIP Server +-----+ +-+ +------+ 66.1.2.3--internet--208.4.5.6|Cisco|----10.1.1.2|S|10.2.1.1--.2|PSTN | | 1:1 | |E| |Switch| 208.4.5.7| NAT |----10.5.1.2|R|10.6.1.1--.2|TDM | +-----+ +-+ +------+ RTP RTP RTP 208.4.5.6 is converted to 10.1.1.2 and vice versa 208.4.5.7 is converted to 10.5.1.2 and vice versa (Cisco command) ip nat inside source static 10.1.1.2 208.4.5.6 ip nat inside source static 10.5.1.2 208.4.5.7 rtpproxy is in use, and has 10.5.1.2(em5) (toward Cisco) and 10.6.1.1(em4) (toward PSTN switch). SER listens on 10.1.1.2(em1) and 10.2.1.1(em0). Ser.cfg takes care of the small amount of SIP-related tweaking that must be done to make SIP work: if (dst_ip==10.1.1.2) { #Received on em1 force_send_socket("10.2.1.1"); #emit on em0 rewritehost("10.2.1.2"); #Send to PSTN Switch $gw_ip = "10.2.1.2"; #Send to PSTN Switch When the INVITE message finally reaches the PSTN switch, it says: U 2008/03/23 04:05:55.780069 10.2.1.1:5060 -> 10.2.1.2:5060 INVITE sip:4097679102@10.2.1.2 SIP/2.0. Via: SIP/2.0/UDP 10.2.1.1;branch=z9hG4bK7572.3545b131.0. Via: SIP/2.0/UDP 66.1.2.3:5060;branch=z9hG4bK39eee695;rport=5060. From: "VOIP TEST 1000" <sip:4697271700@66.1.2.3>;tag=as571000e7. To: <sip:4097679102@208.4.5.6>. Contact: <sip:4697271700@66.1.2.3>. Call-ID: 2bf456fe3016fd6b62a1649c7cb6cd75(a)66.1.2.3. CSeq: 102 INVITE. User-Agent: Asterisk PBX. Max-Forwards: 16. Date: Sun, 23 Mar 2008 04:05:50 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Supported: replaces. Content-Type: application/sdp. Content-Length: 490. . v=0. o=root 55522 55522 IN IP4 66.1.2.3. s=session. c=IN IP4 10.6.1.1. t=0 0. m=audio 35010 RTP/AVP 0 3 8 112 5 10 7 97 111 101. a=rtpmap:0 PCMU/8000. a=rtpmap:3 GSM/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:112 AAL2-G726-32/8000. a=rtpmap:5 DVI4/8000. a=rtpmap:10 L16/8000. a=rtpmap:7 LPC/8000. a=rtpmap:97 iLBC/8000. a=fmtp:97 mode=30. a=rtpmap:111 G726-32/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. a=sendrecv. So the invite sent to the PSTN looks wonderful, correctly pointing SIP and RTP back to the SER box. The problem is on the Cisco side of the SER box, and shows up when the 183/200 messages are emitted by the SER. SER/rtpproxy rewrite the c=IN IP4 line to point to 10.5.1.2, which would be fine except that I need the public IP address 208.4.5.7 to appear in this position. I have tried using force_rtp_proxy("flags","208.4.5.7") in the reply_route section, and I know it is getting executed, but nothing changes. I've tried a variety of flag combinations, but obviously haven't tried all 720 combinations (although it feels like I have). I also tried fix_nated_sdp("2") by itself, with an empty force_rtp_proxy() and a force_rtp_proxy("","208.4.5.7") and all that did was create a "none of the above value" for c= of 10.6.1.110.5.1.2, pasting the two IP addresses together from the rtpproxy command line, and ignoring the value right value I'm trying to provide. So, the question is: How do you say to SER "Here is the value that should be used for c= on this call"? How do I set this? For complex reasons eliminated here to make the example simple, extending the public IP address over to the em5 interface so it is public already is not practical. I need a fix that changes the SDP c= value to one I will know based on whom the distant asterisk server (or equivalent) device will be. There may be other fields that also need fixing for RFC compliancy, but c= being wrong is the current problem. Thanks for suggestions.

1 0

[OpenSER-Users] unixODBC and SQLite???
by Douglas Garstang 21 Mar '08

21 Mar '08

/etc/odbc.ini: [sqlite1] Description = SQLite Driver = SQLite Database = /root/test.db Timeout = 9000 Trace = Yes TraceFile = /tmp/sqlite.log /etc/odbcinst.ini: [SQLite] Description = SQLite ODBC Driver Driver = /usr/local/lib/libsqlite3odbc.so Setup = /usr/local/lib/libsqlite3odbc.so openser.cfg: modparam("avpops", "db_url", "unixodbc://localhost/sqlite1") What's the correct format for specifying the DSN when using SQLite with ODBC? SQLite doesn't use a user/password or host. I tried the above, "unixodbc://localhost/sqlite1" and openser barfs with: ERROR:unixodbc:new_connection: failed to connect I also tried: modparam("avpops", "db_url", "unixodbc://sqlite1") and openser barfed with: new_db_id: Error while parsing database URL: unixodbc://sqlite1 What's the correct format? Is it even supported in Openser 1.2.0? Thanks, Doug ____________________________________________________________________________________ Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping

2 1

[OpenSER-Users] AVP_TABLE.. Why is it required?
by Douglas Garstang 21 Mar '08

21 Mar '08

I put this in openser.cfg: modparam("avpops","avp_url","mysql://unixodbc:passwd@host/database") When I start openser it complains and then EXITS. ERROR:avpops_init: "AVP_DB" present but "AVP_TABLE" found empty Why do I need to specify an avp_table? I plan to use avp_db_query() to retrieve values from another table. That's if using avp_db_query against an odbc source is even possible. People say it is, but given that it's not documented, I'm not sure. Doug. ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ

2 1

[OpenSER-Users] AVPops avp_url???
by Douglas Garstang 21 Mar '08

21 Mar '08

I'm trying to use avpops with unixodbc. What's the format of the avp_uri for unixodbc? The documentation has: modparam("avpops","avp_url","mysql://user:passwd@host/database") What format do I use? Is it this? Is it something else? modparam("avpops","avp_url","unixodbc://user:passwd@host/database") Why isn't this documented? Doug ____________________________________________________________________________________ Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping

2 1

[OpenSER-Users] O'SER behind http Proxy which requires username and Password
by luzango mfupe 21 Mar '08

21 Mar '08

Hi Mates, IM having my OpenSER in the university campus where all the communication have to go through a http Proxy which always asks for user name and password. Where in the O'SER config file should i place these credentials ( http://user:pass@proxyname:port";) Do i have to do the same in the Mediaproxy config file as well? WBR, LU.

2 1

[OpenSER-Users] UA Get's Disconnected Once a call is initiated
by Ali Jawad 21 Mar '08

21 Mar '08

Dear All I am using Openser with TLS, the client connects successfully however once I press the call button the connection to the client get's closed "Actually the client stays open however I do check with netstat and the connection is not there anymore". As for the openser server The following is noted in the logs Mar 18 15:13:56 [1396] DBG:core:handle_ser_child: read response= b613c8d8, 1, fd -1 from 5 (1378) Mar 18 15:13:56 [1378] DBG:core:tcp_send: after receive_fd: c= 0xb613c8d8 n=4 fd=14 Mar 18 15:13:56 [1378] DBG:core:tcp_send: sending... Mar 18 15:13:56 [1378] DBG:core:tls_update_fd: New fd is 14 Mar 18 15:13:56 [1378] DBG:core:tls_write: write was successful (580 bytes) Mar 18 15:13:56 [1378] DBG:core:tcp_send: after write: c= 0xb613c8d8 n=580 fd=14 Mar 18 15:13:56 [1378] DBG:core:tcp_send: buf= SIP/2.0 408 Request Timeout Via: SIP/2.0/TLS 192.168.0.149:9071;rport=9071;branch=z9hG4bK30231106nh26;received=my.own .public.ip From: 9613041705 <sip:9613041705@server.ip>;tag=30231106 To: 9613041705 <sip:9613041705@server.ip>;tag=22c184cf1a1beaaaab06669b2c48f35d-e4d8 Call-ID: F96E5BE097AAtYP4(a)server.ip CSeq: 2 REGISTER Server: OpenSER (1.3.0-tls (i386/linux)) Content-Length: 0 Warning: 392 xx.yy.yy.yy:443 "Noisy feedback tells: pid=1378 req_src_ip=xx.xx.xx.xx req_src_port=9071 in_uri=sip:87.236.144.13 out_uri=sip:pstn.gw.ip.here:5060 via_cnt==0" Mar 18 15:13:56 [1378] DBG:tm:relay_reply: sent buf=0x816aed8: SIP/2.0 4..., shmem=0xb614fb38: SIP/2.0 4 Mar 18 15:13:56 [1378] DBG:tm:final_response_handler: done Mar 18 15:13:56 [1385] DBG:core:tls_update_fd: New fd is 20 Mar 18 15:13:56 [1385] DBG:core:_tls_read: connection closed cleanly Mar 18 15:13:56 [1385] DBG:core:tcp_read_req: EOF Mar 18 15:13:56 [1385] DBG:core:io_watch_del: io_watch_del (0x813d8c0, 20, -1, 0x10) fd_no=2 called Mar 18 15:13:56 [1385] DBG:core:release_tcpconn: releasing con 0xb613c8d8, state -1, fd=20, id=12 Mar 18 15:13:56 [1385] DBG:core:release_tcpconn: extra_data 0xb614c9f0 Mar 18 15:13:56 [1396] DBG:core:handle_tcp_child: reader response= b613c8d8, -1 from 0 Mar 18 15:13:56 [1396] DBG:core:tcpconn_destroy: destroying connection 0xb613c8d8, flags 0002 Mar 18 15:13:56 [1396] DBG:core:tls_close: closing SSL connection Mar 18 15:13:56 [1396] DBG:core:tls_update_fd: New fd is 24 Mar 18 15:13:56 [1396] DBG:core:tls_shutdown: shutdown successful Mar 18 15:13:56 [1396] DBG:core:tls_tcpconn_clean: entered Mar 18 15:13:58 [1378] DBG:tm:utimer_routine: timer routine:7,tl=0xb615f734 next=(nil), timeout=3056900000 Mar 18 15:14:01 [1378] DBG:tm:timer_routine: timer routine:2,tl=0xb615f630 next=(nil), timeout=3060 Mar 18 15:14:01 [1378] DBG:tm:wait_handler: removing 0xb615f5e8 from table Mar 18 15:14:01 [1378] DBG:tm:delete_cell: delete transaction 0xb615f5e8 Mar 18 15:14:01 [1378] DBG:tm:wait_handler: done Mar 18 15:14:41 [1378] DBG:presence:msg_watchers_clean: cleaning pending subscriptions Mar 18 15:14:41 [1378] DBG:core:db_free_rows: freeing 0 rows Mar 18 15:14:41 [1378] DBG:presence:msg_presentity_clean: cleaning expired presentity information Mar 18 15:14:41 [1378] DBG:core:db_free_rows: freeing 0 rows With Regards Ali Jawad System Administrator Splendor Telecom (www.splendor.net) Beirut, Lebanon Phone: +961 1 373725 Fax: + 961 1 375554

2 1

Re: [OpenSER-Users] How configure Voice mail for multi users.
by Mark Sayer 21 Mar '08

21 Mar '08

Asif - Despite what the documentation says, I'd drop the context portion of your command. i.e. exten => s-NOANSWER,n,VoiceMail(${EXTEN}) Since you are clearly going to VM for different reasons you can also add an optional switch that will play a different vm-message. Check the docs for more info. Mark At 03:04 a.m. 20/03/2008, you wrote: >Hi All, >i want to configure voice mail on Asterisk 1.4 for multiple users. let >me explain you the scenario. > >i have 10 users with the name of >1000,2000,3000,4000,5000,6000,.......and these user can call to each >other. Now i want to configure separate voice mail box for separate >user. > >my extensions.conf ..... settings below.. >[voicemail] >exten => _X.,1,Dial(SIP/${EXTEN}) >exten => _X.,n,NoOp(Dial Status: ${DIALSTATUS}) >exten => _X.,n,Goto(s-${DIALSTATUS},1) ...........

1 0

[OpenSER-Users] new CDRTool release 6.3.0
by Adrian Georgescu 21 Mar '08

21 Mar '08

The software can be downloaded from: http://download.dns-hosting.info/CDRTool/ Changelog since 6.1.0: cdrtool (6.3.0) unstable; urgency=low * Boost the speed of the rating engine from 100% to 1000% depending on the complexity of the rating tables * Fixed set of delimiter from global.inc setting when exporting csv files * Added GetEntityProfiles command to rating engine * Return json encoded for GetPrepaidHistory * Added script to create partial billing_rates tables * Moved $CDRTool['rating'] settings to $RatingEngine in global.inc * Added $RatingEngine['split_rating_table'] option to query rating data in multiple tables, each table name is autogenerated from the billing_rates.name field * Fixed set of timeout for each SOAP connection * Adjust some column lengths Must apply setup/mysql/alter_tables.mysql cdrtool (6.2.3) unstable; urgency=low * Print information about the normalize lock in syslog and to stdout * Log if rating engine takes more time to complete. Must set $RatingEngine['log_delay'] to a value in micro seconds in global.inc and restart ratingEngine * Small html changes in sip settings page * Renamed RatingEngine object to differ from the same setting name from global.inc cdrtool (6.2.2) unstable; urgency=low * Added setting in global.inc $RatingEngine['prepaid_lock'] to enable multiple parallel calls using the same prepaid account, this is done by disabling the lock (setting it to 0 or false). Disable the lock only if you accept that the balance might become negative * Added email filter to Sip->getAccounts() * Log the call start time in the head of rating info to eliminate the confusion when rating calls that span multiple profiles * Show total call duration in rate info head * Use <voice-mailbox> instead of absolute URIs for call diversions * Removed unnecessary code used in the past for replication of changes to remote soap engine * Fixed replication of customer account to remote platform * Add active_master to replicatedDatabases in global.inc * Added web link between sip target and sip_accounts * Rewrite of mysql monitoring functions * Added DeleteBalance and DeleteBalance History commands to rating engine * Use colors in mysql replication status * Added posibility to replicate customer entries in Customer Port * Added getPrepaidHistory function to the rating engine * Replaced false with 0 and true with 1 for the return values from rating engine * Show database ip address when replication monitor returns an error * Fixed name of sip_ports settings * Added setup/crontabs directory for non debian installations cdrtool (6.2.1) unstable; urgency=low * Correct update of radius records when mediaproxy field is NULL or !timeout You must reload the sql stored procedures from setup/radius/OpenSER/radius_accounting.proc or update Freeradius with setup/radius/OpenSER/sql.conf when not using the stored procedures * Log add balance to prepaid_history table cdrtool (6.1.9) unstable; urgency=low * Fixed memory leak related to calling the same soap function several times * Added separate access numbers for FUNV, FBUS, FNOL, FNOA * Fixed load of customer properties in ngnpro client * Support multiple clusters for mysql replication monitor * Fixed vulnerability related to missing BYE and MediaProxy radius Update action (reported by Inaki Baz Castil) You must reload the sql stored procedures from setup/radius/OpenSER/radius_accounting.proc or update Freeradius with setup/radius/OpenSER/sql.conf when not using the stored procedures * Specify if output of soap functions is html formatted or not cdrtool (6.1.8) unstable; urgency=low * Fixed mrtg buildStatistics call in cron.d (env LANG=C) * Added web page to show replications status and dynamic instructions about how to fix it * Replace $CDRTool['mysql_clusters'] in global.inc with $replicated_databases (see setup/global.inc.in for syntax) * Delete DB1 and DB2 definitions from global.inc cdrtool (6.1.7) unstable; urgency=low * Set mobile_number per SIP account * Add sip_accounts_lite to soap engine settings * Update voicemail full name when sip account firstname or lastname changes * Create templates_c dir required by smarty * Added templates directory for sending welcome note cdrtool (6.1.6) unstable; urgency=low * Fixed load of customer profiles for prepaid calls when no default customer defined cdrtool (6.1.5) unstable; urgency=low * Added lock icon when SIP UA registered using TLS contact * Added more information about renormalizing prepaid calls * Show port number and transport protocol in sip trace * Group trace message per IP instead of ip:port * Added welcome message per soap engine * Show error message when ngnpro client is not configured * Fixed link from sip domains to sip aliases page * Fixed ENUM soap port authentication in sip settings page * Replaced DB_ser with DB_openser Edit global.inc and replace DB_ser with DB_openser * Increased curl timeout for bulk SOAP operations * Return price in debit balance cdrtool (6.1.4) unstable; urgency=low * Remove sample mrtg files cdrtool (6.1.3) unstable; urgency=high * Fixed calculation of rating profiles when using having specific gateways/domains/subscribers * Fixed default port alocation for NGNPro version < 3 * Allow set of reseller 0.0 (admin) in login account * Show SipThor node in Sip settings title bar * Fixed management of presence rules * Fixed link from Sip accounts to sip settings page cdrtool (6.1.2) unstable; urgency=high * Fixed detection of CDR duration when minimumDuration was set cdrtool (6.1.1) unstable; urgency=low * Corrected default login account sql insert statement Regards, Adrian

9 9

[OpenSER-Users] How configure Voice mail for multi users.
by Mian M Asif 20 Mar '08

20 Mar '08

Hi All, i want to configure voice mail on Asterisk 1.4 for multiple users. let me explain you the scenario. i have 10 users with the name of 1000,2000,3000,4000,5000,6000,.......and these user can call to each other. Now i want to configure separate voice mail box for separate user. my extensions.conf ..... settings below.. [voicemail] exten => _X.,1,Dial(SIP/${EXTEN}) exten => _X.,n,NoOp(Dial Status: ${DIALSTATUS}) exten => _X.,n,Goto(s-${DIALSTATUS},1) exten => s-NOANSWER,1,Background(vm-nobodyavail) exten => s-NOANSWER,n,VoiceMail(${EXTEN}@usersmail) exten => s-NOANSWER,n,Hangup() exten => s-CONGESTION,1,Background(vm-nobodyavail) exten => s-CONGESTION,n,VoiceMail(${EXTEN}@usersmail) exten => s-CONGESTION,n,Hangup() exten => s-CANCEL,1,Background(vm-nobodyavail) exten => s-CANCEL,n,VoiceMail(${EXTEN}@usersmail) exten => s-CANCEL,n,Hangup() exten => s-BUSY,1,Background(salesrep) exten => s-BUSY,n,VoiceMail(${EXTEN}@usersmail) exten => s-BUSY,n,Hangup() exten => s-CHANUNAVAIL,1,Background(vm-nobodyavail) exten => s-CHANUNAVAIL,n,VoiceMail(${EXTEN}@usersmail) exten => s-CHANUNAVAIL,n,Hangup() my voicemail.conf [usersmail] 1000 => 1212, userm, 1000(a)abc.com 2000 => 1212, userm, 2000(a)abc.com please help me how can i set calling number before send voice mail in users voicemail box. when i dial like VoiceMail(${EXTEN}@usersmail) voice mail not work and when i heard code user, like this VoiceMail(1000@usersmail) voicemail work fine. but i want to set if user dial 2000 or 3000 or 4000 it should be set automatically. thanks for your cooperations. regards, Asif

4 8

[OpenSER-Users] LCR Problems (1.1 -> 1.2)
by Kristian Kielhofner 20 Mar '08

20 Mar '08

Hello everyone, In OpenSER 1.1 the following route block worked perfectly: route[5] { # LCR PSTN handler xlog("$rm $si $ru $ci - LCR - Starting\n"); if (!load_gws()) { xlog("$rm $si $ru $ci - LCR - Error in load_gw, setting engerr 21\n"); avp_printf("$avp(s:engerr)", "21"); route(15); }; if (!next_gw()) { xlog("$rm $si $ru $ci - LCR - Error in next_gw, setting engerr 22\n"); avp_printf("$avp(s:engerr)", "22"); route(15); }; t_on_failure("5"); xlog("$rm $si $ru $ci - LCR - Sending to $td\n"); route(1); } In OpenSER 1.2.3 it fails on next_gw() every time but I am unable to find out why. I used the new convert option to mysqldb.sh and I think my LCR tables are ok. I tried turning up the verbosity but I couldn't see anything relevant to LCR. What could be going on here? Thanks! -- Kristian Kielhofner

2 2

[OpenSER-Users] SQLite with OpenSER
by Douglas Garstang 20 Mar '08

20 Mar '08

How can I use avp_db_query() with SQLite in OpenSER? Doug. ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ

2 1

[OpenSER-Users] One way audio calling a -> b, two way audio calling b -> a with two phones behind the same NAT
by Mike Fedyk 20 Mar '08

20 Mar '08

Hi, I am having some trouble with nat traversal. Depending on which phone is originating the call, it may either have one way audio, two way, or even neither. Here is the matrix I have tested so far. Hardphone to hard phone works fine between two spa942 or either of the spa942 phones and the budgetone, the problems come into the picture when soft phones are used. caller callee RTP direction budgetone linphone neither budgetone spa942 both budgetone xlite both linphone budgetone one way linphone spa942 one way linphone xlite neither spa942 budgetone both spa942 linphone neither spa942 xlite one way xlite budgetone both xlite linphone both xlite spa942 both Openser.cfg http://pastebin.ca/946561

2 2

[OpenSER-Users] Quality Metrics with MediaProxy
by Douglas Garstang 20 Mar '08

20 Mar '08

Question.... If I was to use mediaproxy with OpenSER, is there any way to capture RTP quality metrics with it? Thanks, Doug. ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ

1 0

[Serusers] General Question on avp_subst
by Mike Trest - Personal 20 Mar '08

20 Mar '08

Hi, I am confused by documentation with foo bar examples. Can someone please give me a clear ser.cfg example of avp_subst that really works in your running ser.cfg? preferably one that works on and avp called $revision to substitute a substring of numbers (8831) with another set of numbers (5431) where the target digits are surrounded by other stuff. ..mike.. .

1 1

Re: [OpenSER-Users] openser msilo problem
by Daniel-Constantin Mierla 20 Mar '08

20 Mar '08

Hello, On 03/18/08 13:02, Devesh wrote: > Hi Daniel, > Again thanks. > I have commented authentication part from file openser.cfg . But this is not > working for me . there might be other problems. > Where should I change in file. > that is hard to tell :-) I would recommend you to install ngrep, wireshark or other network sniffing tool and watch the sip traffic. It will help to troubleshoot better, you will see where sip messages are sent and where (if) are blocked. Cheers, Daniel PS. please cc to mailing list. > Thanks & Regards > Devesh Bissa > persistent system ltd. > > > > -----Original Message----- > From: Daniel-Constantin Mierla [mailto:miconda@gmail.com] > Sent: Tuesday, March 18, 2008 3:27 PM > To: Devesh > Cc: users(a)openser.org > Subject: Re: [OpenSER-Users] openser msilo problem > > Hello, > > seems that you authenticate the MESSAGEs sent by msilo -- filter out > these ones, the source IP should be localhost. > > Cheers, > Daniel > > > On 03/18/08 11:49, Devesh wrote: > >> Hi Daniel, >> Thanks for your reply. >> I am not able to grep any message because this utility is not on my >> > machine. > >> I am sending you MSILO log.(attached) >> Please guide me where I am wrong. >> >> >> Thanks & Regards >> Devesh Bissa >> persistent system ltd. >> >> >> >> -----Original Message----- >> From: Daniel-Constantin Mierla [mailto:miconda@gmail.com] >> Sent: Tuesday, March 18, 2008 2:30 PM >> To: Deveshb >> Cc: users(a)lists.openser.org >> Subject: Re: [OpenSER-Users] openser msilo problem >> >> Hello, >> >> On 03/14/08 16:17, Deveshb wrote: >> >> >>> Hi, >>> I have install openser with msilo module . Offline messages stored >>> >>> >> there >> >> >>> but it's not deliver to client( I am using X-Lite). >>> it gives me following error >>> DBG:maxfwd:is_maxfwd_present: max_forwards header not found! >>> DBG:uri:has_totag: no totag >>> DBG:tm:t_lookup_request: start searching: hash=3841, isACK=0 >>> DBG:tm:matching_3261: RFC3261 transaction matching failed >>> DBG:tm:t_lookup_request: no transaction found >>> >>> >>> >> these are not errors, are debug messages. >> >> Grab the SIP traffic for this case and and send here, that will show >> where the message is delivered. Sample command: >> >> ngrep -d any -qt -W byline port 5060 >> >> Cheers, >> Daniel >> >> >> >> >> >>> Could you please help me . >>> >>> Thank you >>> Devesh >>> >>> >>> >> DISCLAIMER >> ========== >> This e-mail may contain privileged and confidential information which is >> > the property of Persistent Systems Ltd. It is intended only for the use of > the individual or entity to which it is addressed. If you are not the > intended recipient, you are not authorized to read, retain, copy, print, > distribute or use this message. If you have received this communication in > error, please notify the sender and delete all copies of this message. > Persistent Systems Ltd. does not accept any liability for virus infected > mails. > >> >> > > > DISCLAIMER > ========== > This e-mail may contain privileged and confidential information which is the property of Persistent Systems Ltd. It is intended only for the use of the individual or entity to which it is addressed. If you are not the intended recipient, you are not authorized to read, retain, copy, print, distribute or use this message. If you have received this communication in error, please notify the sender and delete all copies of this message. Persistent Systems Ltd. does not accept any liability for virus infected mails. >

2 2

[OpenSER-Users] allow_trusted evaluates to false in 1.3.1
by CSB 20 Mar '08

20 Mar '08

I have the following permissions module setup: modparam("auth_db|permissions|uri_db|usrloc|domain|lcr", "db_url", "mysql://user:password@localhost/ser") modparam("permissions", "db_mode", 0) modparam("permissions", "trusted_table", "trusted") if (!is_uri_host_local()) { if (is_from_local() || allow_trusted() ) { route(5); route(1); } else { sl_send_reply("403", "Forbidden"); xlog("403 in call type processing. rs $rs si $si rm $rm ru $ru tu $tu fu $fu fd $fd rr $rr"); }; return; }; When an invite is received the calls is rejected with a 403 and "403 in call type processing" is written to the log file. So allow_trusted must be evaluating to false I can see from the log that the following query is executed: 13 Query select proto,from_pattern,tag from trusted where src_ip='202.180.001.001' When I execute this query directly on the database it returns: +-------+--------------+------+ | proto | from_pattern | tag | +-------+--------------+------+ | any | NULL | NULL | +-------+--------------+------+ 1 row in set (0.00 sec) Could anyone advise what I've done wrong? OpenSER1.3.1 Regards Cameron

1 0

[Serusers] Memory Leak Problems
by Ricardo Martinez 19 Mar '08

19 Mar '08

Hello I'm using SER with the LCR module and i'm having some "memory leak" problems. I have followed the "Problem Reporting" link in the iptel.org site to send you a report. My question is : In the page you mention that i need to use memlog=2 and debug=3. Is this ok? or i need to run SER with more debug to see the problem and report it? Second. How do i know if i'm using the last stable, bug free (in theory) version of SER?. SER -V : version: ser 0.9.6 (i386/linux) flags: STATS: Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 @(#) $Id: main.c,v 1.197.2.1 2005/07/25 16:56:24 andrei Exp $ main.c compiled on 07:54:57 Feb 29 2008 with gcc 4.1.2 i have download this version from the FTP site. I hope someone can help me here. Thanks in advance. Regards, Ricardo Martinez.-

2 4

[OpenSER-Users] openxcap with openser1.2 and postgresql
by Marcelo Rodrigues Leite 19 Mar '08

19 Mar '08

Hello all, I´ve searched around forums and openser and openxcap documentation but I haven´t found any consistent information about the use of openxcap with openser 1.2 and with postgresql. Is it possible to use the 3 of them alltogether? I can overcome any problem with openser 1.2 by installing a newer version, but any mysql restriction is going to be a real problem since I´m using Voip4all architecture, which is built over postgresql. I would really appreciate any information about this. Regards, Marcelo R. Leite

5 5

[OpenSER-Users] About TLS
by Ali Jawad 18 Mar '08

18 Mar '08

Hey All I am trying to setup TLS to use on my openser server this setup allows me to sign in /* uncomment the following lines to enable TLS support (default off) */ #disable_tls = yes disable_tls = no listen = tls:xx.xx.xx.xx:443 tls_verify_server = 1 tls_verify_client = 1 tls_require_client_certificate = 0 tls_method = TLSv1 tls_certificate = "/usr/local/etc/openser/tls/user/user-cert.pem" tls_private_key = "/usr/local/etc/openser/tls/user/user-privkey.pem" tls_ca_list = "/usr/local/etc/openser/tls/user/user-calist.pem" /* uncomment and configure the following line if you want openser to bind on a specific interface/port/proto (default bind on all available) */ listen=udp:xx.xx.xx.xx:5060 However removing listen=udp:xx.xx.xx.xx:5060 blocks me from signing in at all. I have checked with the first configuration that I sign in to port 443, however even if I am able to sign in using port 443 I can make calls I always get time out errors from my openser server. It works fine if I disable tls and only use UDP. Any hints ? Thx

2 1

[OpenSER-Users] INVITE issue - 483 Too Many Hops
by Eloi Pecharroman 18 Mar '08

18 Mar '08

Hello all, I have an issue with my OpenSER setup. I am struggling to fix it and to find further information. I am working with the debian package 1.3.0-tls Registration (with and without authentication works fine), however, when a client sends an INVITE, it seems to be looping back the request until the max-forward is reached and hence returning the error. The log throws out an insane amount of *repeated* requests. My configuration file is the standard one, I have only added a few xlogs to see if the uri matched my alias, which strangely seems to do as "URI is not mine" is never output. alias=sipserver [....] if(uri!=myself) { xlog("L_NOTICE","URI is not mine\n"); route(1); exit; } [...] route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; exit; } Eloi epecharroman(a)aptx.com

2 1

Re: [OpenSER-Users] openser msilo problem
by Daniel-Constantin Mierla 18 Mar '08

18 Mar '08

Hello, seems that you authenticate the MESSAGEs sent by msilo -- filter out these ones, the source IP should be localhost. Cheers, Daniel On 03/18/08 11:49, Devesh wrote: > Hi Daniel, > Thanks for your reply. > I am not able to grep any message because this utility is not on my machine. > I am sending you MSILO log.(attached) > Please guide me where I am wrong. > > > Thanks & Regards > Devesh Bissa > persistent system ltd. > > > > -----Original Message----- > From: Daniel-Constantin Mierla [mailto:miconda@gmail.com] > Sent: Tuesday, March 18, 2008 2:30 PM > To: Deveshb > Cc: users(a)lists.openser.org > Subject: Re: [OpenSER-Users] openser msilo problem > > Hello, > > On 03/14/08 16:17, Deveshb wrote: > >> Hi, >> I have install openser with msilo module . Offline messages stored >> > there > >> but it's not deliver to client( I am using X-Lite). >> it gives me following error >> DBG:maxfwd:is_maxfwd_present: max_forwards header not found! >> DBG:uri:has_totag: no totag >> DBG:tm:t_lookup_request: start searching: hash=3841, isACK=0 >> DBG:tm:matching_3261: RFC3261 transaction matching failed >> DBG:tm:t_lookup_request: no transaction found >> >> > these are not errors, are debug messages. > > Grab the SIP traffic for this case and and send here, that will show > where the message is delivered. Sample command: > > ngrep -d any -qt -W byline port 5060 > > Cheers, > Daniel > > > > >> Could you please help me . >> >> Thank you >> Devesh >> >> > > DISCLAIMER > ========== > This e-mail may contain privileged and confidential information which is the property of Persistent Systems Ltd. It is intended only for the use of the individual or entity to which it is addressed. If you are not the intended recipient, you are not authorized to read, retain, copy, print, distribute or use this message. If you have received this communication in error, please notify the sender and delete all copies of this message. Persistent Systems Ltd. does not accept any liability for virus infected mails. >

1 0

[OpenSER-Users] interconnect with another operator trough sip
by Ion Minzu 18 Mar '08

18 Mar '08

Hi, The architecture looks like this: PSTN <-> AS5300 <-> openser(non_tls) <-> IP-PBX's (asterisk) ^ ^ ^ billing^ (radius) i want to interconnect with another operator trough sip. i think about two methods: -using AS5300 -directly to openser. what method do you recommend for me? and how do you think it's resonable to instal another openser with tls wich will function like session border controller for interconnections with operators through sip. The goal is to not change the architecture of billing and existing SIP proxy. i've read something on the sipforum but i am interested if someone has some experience in such cases, taking into consideration that fact that operators use different equipment that supports sip. i mean that i want in result something like this: | | PSTN <-|-> AS5300 <-> | ^ openser(non_tls) <-> IP-PBX's (asterisk) SIP <-|-> openser<-> ^ | ^ ^ | ^ billing^ (radius)

2 1

[OpenSER-Users] PAI-Question
by Helmut Kuper 18 Mar '08

18 Mar '08

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, is there a way to access two or more P-Asserted-Identity header field directly with PVs? The only way I found to do is via $(hdr(P-Asserted-Identity)[N]). I ask because of performance ... regards helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFH3rA+4tZeNddg3dwRAl6xAJ4p4Tv+ih+m46yWB6GRUZuU1wS/8wCaAlOY DtCykNG42Im8cyhVQ1jn2RE= =tKfw -----END PGP SIGNATURE-----

4 3

[OpenSER-Users] openser msilo problem
by Deveshb 18 Mar '08

18 Mar '08

Hi, I have install openser with msilo module . Offline messages stored there but it's not deliver to client( I am using X-Lite). it gives me following error DBG:maxfwd:is_maxfwd_present: max_forwards header not found! DBG:uri:has_totag: no totag DBG:tm:t_lookup_request: start searching: hash=3841, isACK=0 DBG:tm:matching_3261: RFC3261 transaction matching failed DBG:tm:t_lookup_request: no transaction found Could you please help me . Thank you Devesh -- View this message in context: http://www.nabble.com/openser-msilo-problem-tp16048145p16048145.html Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

2 1

[OpenSER-Users] Error openser + freeradius
by MagoDaViola 18 Mar '08

18 Mar '08

Untitled DocumentHi. I´m using a server with OpenSER + MySql FreeRadius + + MediaProxy, and it´s working well, but whenever I get this error: / Usr / local / sbin / openser [2610]: ERROR: auth_radius: radius_authorize_sterman: rc_auth failed All customers are authenticated and holding calls. What can that be? Alcindo Schleder.

2 1

[Serusers] ser_ctl and PostgreSQL backend
by Marcus Better 18 Mar '08

18 Mar '08

Hi, I'm trying to use SER 2.0 (daily snapshot) with a PostgreSQL backend, and have a few problems. What is the status of PostgreSQL support in SER? I created the schema with the ser_postgres script. That didn't populate the version table, so I had to do that manually. Now I need to add a domain with ser_ctl, but get this: ERROR: argument of AND must be type boolean, not type integer STATEMENT: SELECT did FROM domain WHERE 1 AND (NOT ((flags & 128)!='0')) AND (did='example.org') LIMIT 1 Is there another way to create domain entries, or should I switch to MySQL? Regards, Marcus Better

3 3

[OpenSER-Users] Your Industry profile
by VOIP Expert 18 Mar '08

18 Mar '08

1 0

[OpenSER-Users] Your Industry profile
by VOIP Expert 18 Mar '08

18 Mar '08

1 0

[OpenSER-Users] Relaying to outbound proxy
by Adrian A 17 Mar '08

17 Mar '08

Hi, I'm setting up multiple OpenSER proxies in different locations to server a SIP domain. The proxies use a shared location database so they always know where to send requests to the client when an incoming call hits either one. The clients have the outbound proxy set to whichever one is closest to them. The problem is that because the clients are NAT they do not receive the request if the proxy IP that sends it is not the same as the one they have set as the outbound proxy. The NAT will reject the request because it does not arrive from the same IP that the connection was established to. How can I make OpenSER relay requests to the "home proxy" of the user so that this can then forward it to the client? I have added an "ob_proxy" column to the subscriber table and tried using avp_db_load to get this value for use in a t_relay() but I'm getting errors in OpenSER. With the following config: modparam("avpops","avp_url","mysql://user:pass@localhost/openser") modparam("avpops", "avp_table", "dummy") modparam("avpops","db_scheme", "scheme1:username_col=username;value_col=ob_proxy;table=subscriber") avp_db_load("$to","$avp(s:ob_proxy)/subscriber"); I'm getting errors in Openser log: /usr/local/sbin/openser[15470]: ERROR:mysql:db_mysql_submit_query: driver error: Unknown column 'value' in 'field list' /usr/local/sbin/openser[15470]: ERROR:mysql:db_mysql_query: error while submitting query /usr/local/sbin/openser[15470]: ERROR:avpops:ops_dbload_avps: db_load failed Does anyone know how to fix this or is there a better way of doing it?

3 8

[OpenSER-Users] Cant connect to local MySQL server through socket
by Megha Saini 17 Mar '08

17 Mar '08

Hi, I have installed OpenSER, openser-1.3.1-notls from: http://openser.org/pub/openser/1.3.1/src/ on Fedora Core 8. I have followed the steps given in: http://www.openser.org/dokuwiki/doku.php/install:openser-from-svn For the make command, I used: make install include_modules="mysql" I have completed all the steps successfully. However, I am not able to create database as said in step 7. To check that MySQL is working on my system, I used the command: /etc/init.d/mysqld start It says: Starting MySQL : [OK] To check openser is running: I used the command, /etc/init.d/openser start I get on command line: Starting openser: [OK] However, when i try to create database: /usr/local/sbin/openserdbctl create I get the error as: database engine 'mysql' loaded INFO: test server charset ERROR 2002 (HY000): Can't connect to local MySQL server through socket '/var/lib/mysql/mysql.socket' (2) ERROR 2002 (HY000): Can't connect to local MySQL server through socket '/var/lib/mysql/mysql.socket' (2) Usage: egrep [OPTION]... PATTERN [FILE]... Try `egrep --help' for more information. /usr/local/lib/openser/openserctl/openserdbctl.mysql: line 112: [: =: unary operator expected INFO: creating database openser ... ERROR 2002 (HY000): Can't connect to local MySQL server through socket '/var/lib/mysql/mysql.socket' (2) ERROR: Creating core database and grant privileges failed! I checked the file mysql.socket in /var/lib/mysql, there is only an empty file with this name. As seen in forums, I saw that many people have this file in /tmp/ folder. However, I dont have this file there also! I am unable to create the database and therefore I cannot register the client using X-lite or SJPhone. Kindly help me out with this. I am badly stuck with this problem since 3 days and have been looking in forums a lot. I am not getting any clue of how to proceed further. Please help me out if anybody has any clue. I look forward to this mailing list as my only help. I need to proceed further as I have lot of things ot be done in this project and my deadline is very near. Looking forward to your co-operation. Thanks in advance. Megha

3 3

[OpenSER-Users] R: Error in pua_usrloc: PUBLISH with empty body (bug 1821002)
by Zappasodi Daniele 17 Mar '08

17 Mar '08

Hi, thanks for the reply. I haven't found an expires paramater of usrloc module, maybe you mean the expires of registrar module: modparam("registrar", "max_expires", 3600) If it is so, I have tried with the same value for presence and registrar modparam("presence", "max_expires", 3600) and with presence expires greater than the registrar expires modparam("presence", "max_expires", 4000) I have repeted the test and I have seen that the error messages occur when I deregister and reregister the subscribed client, but I think that the error could be related to the fact that the subscriber (Eyebeam) refuses the first NOTIFY (why?) sended by openser for the first REGISTER from the subscribed. In the attached file you can see the syslog with debug level = 9 ********************************************************************** The information in this message is confidential and may be legally privileged. It is intended solely for the addressee. Access to this message by anyone else is unauthorized. If you are not the intended recipient, any disclosure, copying, or distribution of the message, or any action or omission taken by you in reliance on it, is prohibited and may be unlawful. Please immediately contact the sender if you have received this message inerror. **********************************************************************

1 0

[Serusers] MAC client
by jiri＠iptel.org 16 Mar '08

16 Mar '08

any recommendation for a good Mac client? Thanks! -jiri

3 2

[OpenSER-Users] SIP/2.0 481 Call Leg/Transaction Does Not Exist
by shiebar＠post.tau.ac.il 14 Mar '08

14 Mar '08

Hi, I am new with openser. I was able to configure openser with wesip that both will run. When trying to make a call between two softphone I get "SIP/2.0 481 Call Leg/Transaction Does Not Exist". Both caller and callee are registered. In addition I see that after creating the db I don't have a user openser with read-write permmissions. > openserctl acl show openser Non-existent user 'openser'. Still proceeding? [Y|N] I have used "openserdbctl create" to create the db. And i got no error message when creating it. WeSIP get the message with the openser IP and not with the callee IP. Could it be the reason for the 481 reply? Thanks in advance, Shie

1 0

[Serusers] Moved Temporarily 302 - redirect server
by Ladislav Andel 14 Mar '08

14 Mar '08

Hi all, can I with SER add following in 302 response? 1) add a parameter in Contact header that a UA will read and copy it to the new INVITE and how would I do it in ser config. 2) or add a new header field with in 302 that a UA will read and copy it to the new INVITE The first should be possible as RFC3261 in 20.10 shows in example but I'm wondering if SER can do it too. Thank you, Lada

4 8

[OpenSER-Users] Acceptor peer failed establishing state i came to know about this list today and i thought to send problem. if anybody know the solution or have any idea please do let me know. actually i am authenticating a user using openser and opendiameters NASREQ applica
by bhargav.b＠renovau.com 14 Mar '08

14 Mar '08

hi all, i came to know about this list today and i thought to send problem. if anybody know the solution or have any idea please do let me know. actually i am authenticating a user using openser and opendiameters NASREQ application as the client and server.at present the openser is sending the all needed avp's to opendiameter client.i can see all the avp code and avp data on the client side. my problem is that the on the diameter server side the avp,s are not transferred by the client. the client is giving something like (6372|1698151312) Acceptor peer failed establishing state [3003], closing i am confused about it.before i sent the avp's from openser side there was no such problem. seeking help regards

1 0

[OpenSER-Users] SIP/2.0 481 Call Leg/Transaction Does Not Exist
by bsb093 14 Mar '08

14 Mar '08

Hi, http://www.nabble.com/file/p16047318/WeSIP-debug WeSIP-debug I an trying to make a call between 2 softphones. If I try to make the call directly there is no problem but when trying to use openser with wesip as the server I get: SIP/2.0 481 Call Leg/Transaction Does Not Exist I have tried to check it with the debug but I can not find the problem. OpenSER replace the To header uri with the correct uri: rewrite_uri: rewriting Request-URI with 'sip:shine_sip2@192.168.2.101:5070' Attached are the OpenSER and WeSIP debug output. I will just add that the call is from shine_sip(192.168.2.102) to shine_sip2(192.168.2.101) and if I check openser for online user I see that both users are online: dell110l:/home/mil# openserctl ul show Domain:: aliases table=512 records=0 max_slot=0 Domain:: location table=512 records=3 max_slot=1 AOR:: shine_sip2 Contact:: sip:shine_sip2@192.168.2.101:5070 Q=0.9 Expires:: 445 Callid:: 1205472816-140-HOME-7C77A26D29(a)192.168.2.101 Cseq:: 97 User-agent:: NCH Swift Sound Express Talk 3.06 State:: CS_NEW Flags:: 0 Cflag:: 0 Socket:: udp:192.168.2.100:5060 Methods:: 4294967295 AOR:: shine_sip Contact:: sip:shine_sip@10.108.3.84:5070 Q=0.9 Expires:: 455 Callid:: 1205473792-3532-BSB093-31(a)192.168.2.102 Cseq:: 63 User-agent:: NCH Swift Sound Express Talk 3.06 State:: CS_NEW Flags:: 0 Cflag:: 0 Socket:: udp:192.168.2.100:5060 Methods:: 4294967295 Contact:: sip:shine_sip@10.108.3.84:5061 Q=0.9 Expires:: 1204 Callid:: 1205473792-3532-BSB093-31(a)192.168.2.102 Cseq:: 64 User-agent:: NCH Swift Sound Express Talk 3.06 State:: CS_NEW Flags:: 0 Cflag:: 0 Socket:: udp:192.168.2.100:5060 Methods:: 4294967295 Please advice. Thanks, Shine http://www.nabble.com/file/p16047318/OpenSER-debug OpenSER-debug -- View this message in context: http://www.nabble.com/SIP-2.0-481-Call-Leg-Transaction-Does-Not-Exist-tp160… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

1 0

Re: [OpenSER-Users] Errors
by Bai Shi 14 Mar '08

14 Mar '08

Hi TA, Please check ur config file line 163 as mentioned in the debug log: CRITICAL:core:yyerror: parse error in config file, line 163, column 1-9: Invalid arguments Mar 11 22:36:20 [3449] If you have any problem with it, you can try to post ur config file in the mailing list, so other may be able to help you. Rgds, BS

2 3

[OpenSER-Users] Preserve the original number with alias_db_lookup
by Morten Isaksen 13 Mar '08

13 Mar '08

Hi! I have some challenges with alias_db_lookup and hope someone here can help. I have a PBX that has registered 111@xxx with my OpenSER. I want to attach 222@xxx and 333@xxx to that registration and alias_db_lookup does that fine. If a call comes to 111 the sip header looks like this INVITE 111@xxx .... To: 111 <111@xxx> ... And everything works fine. But if a call comes to 222 the sip header looks like this: INVITE 111@xxx .... To: 222 <222@xxx> ... But now the PBX thinks that the call is for 111 and not 222. On Asterisk I can do something like this to solve the problem: exten => _X.,1,goto(from-siptrunk,${CUT(CUT(SIP_HEADER(To),@,1),:,2)},1) [from-siptrunk] exten => 111,1,dial(SIP/1010) exten => 222,1,dial(SIP/1020) But a lot of SIP enabled PBX's cannot do that sort of thing. One solution is the force the PBX to register with OpenSER for every number, but I would prefer not to do that. Another solution is to manually populate the alias tables but this is not very elegant. -- Morten Isaksen http://www.misak.dk/blog/

2 2

Re: [OpenSER-Users] OpenSER + OpenXcap for pres-rules: no NOTIFY
by Mircea Amarascu 13 Mar '08

13 Mar '08

2 1

[OpenSER-Users] Error in pua_usrloc: PUBLISH with empty body (bug 1821002)
by Zappasodi Daniele 13 Mar '08

13 Mar '08

Hello, what about the bug 1821002 on pua_usrloc: PUBLISH with no body? I read that is closed, but in the release 1.3.1 it is still present. Mar 13 14:15:12 localhost ./openser[16477]: ERROR:pua:send_publish: New PUBLISH and no body found- invalid request Mar 13 14:15:12 localhost ./openser[16477]: ERROR:pua_usrloc:ul_publish: while sending publish ********************************************************************** The information in this message is confidential and may be legally privileged. It is intended solely for the addressee. Access to this message by anyone else is unauthorized. If you are not the intended recipient, any disclosure, copying, or distribution of the message, or any action or omission taken by you in reliance on it, is prohibited and may be unlawful. Please immediately contact the sender if you have received this message inerror. **********************************************************************

2 1

[OpenSER-Users] Is it a bug? TM module is sending CANCEL request regardless what the script do with the From Header since 1.3.0
by Bai Shi 13 Mar '08

13 Mar '08

I'm not sure if it is a bug. Currently our set up need to do masquerading on the From header for all invite requests. It used to work okie when we process for the CANCEL request in the routing logic. But after upgrading to openser 1.3.0, the TM module is sending CANCEL out using local cached From info rather than the info processed by uac_replace_from. May I know is this by design or it is a bug? Rgds, BS

3 12

[Serusers] serweb fifo error
by Adeyeye O. Michael 13 Mar '08

13 Mar '08

How can I fix this error? WARNING: no fork mode stateless - initializing 0(0) Maxfwd module- initializing 0(0) INFO: udp_init: SO_RCVBUF is initially 110592 0(0) INFO: udp_init: SO_RCVBUF is finally 221184 2(0) INFO: fifo process starting: 6710 2(6710) SER: open_uac_fifo: fifo server up at /tmp/ser_fifo... 2(6710) ERROR: open_reply_pipe: open error (/tmp/webfifo_1171263842): No such file or directory 2(6710) ERROR: fifo_reply: no reply pipe /tmp/webfifo_1171263842 2(6710) ERROR: open_reply_pipe: open error (/tmp/webfifo_1374470966): No such file or directory 2(6710) ERROR: fifo_reply: no reply pipe /tmp/webfifo_1374470966 2(6710) ERROR: open_reply_pipe: open error (/tmp/webfifo_1359801488): No such file or directory 2(6710) ERROR: fifo_reply: no reply pipe /tmp/webfifo_1359801488 2(6710) ERROR: open_reply_pipe: open error (/tmp/webfifo_90028122): No such file or directory 2(6710) ERROR: fifo_reply: no reply pipe /tmp/webfifo_90028122 2(6710) ERROR: open_reply_pipe: open error (/tmp/webfifo_1698796794): No such file or directory 2(6710) ERROR: fifo_reply: no reply pipe /tmp/webfifo_1698796794 ------------------------------------------------------------------------------------------------ Below is an excerpt of my ser.cfg ------------------------------------------------------------------------------------------------ # $Id: hello-world.cfg 46 2006-01-25 04:21:30Z /CN=Paul Hazlett/emailAddress=paul(a)onsip.org $ debug=3 fork=no log_stderror=yes listen=192.168.0.2 port=5060 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_mode = 0666 fifo_db_url="mysql://ser:heslo@localhost/ser" --------------------------------------------------------------------------------------------- Yours, Mike.

1 0

[OpenSER-Users] UDP messages size
by Ariadne Ramos 12 Mar '08

12 Mar '08

Hello We're having an issue with the INVITE'S size. The INVITE messages are trunked to 1480 Bytes when they exceed that size. Capturing with ethereal, the package appears like "Fragmented IP protocol" (proto=UDP 0x11, off=0) because is incomplete. Exists a package maximum size?, Can I change it?... If so, Can you tell me how to do this? Could be changing max_len keyword? Ariadne L. Ramos

2 1

[OpenSER-Users] when are AVPs in radius extra evaluated?
by jh＠tutpro.com 12 Mar '08

12 Mar '08

if i put in radius_extra "Some-Known-Attribute=$avp(s:bar)" and in onreply_route if (is_method("INVITE") && t_check_status("200")) { xlog("L_INFO", "Setting <s:bar> to <test>\n"); $avp(s:bar) = "test"; }; i do not get Some-Known-Attribute = "test" to accounting Start request although i do get Mar 12 07:20:32 rautu /usr/sbin/openser[7440]: Setting <s:bar> to <test> in syslog. i didn't find from acc/README information on when the AVPs in radius_extra are evaluated. accounting Start is sent out AFTER final response (200 ok) had been received, because it contains SIP-Response-Code = 200 why doesn't setting $avp(s:bar) in onreply_route cause Some-Known-Attribute to be included in accounting start record? -- juha

2 2

[OpenSER-Users] New Soap-Simple-Proxy release 1.0.3
by Adrian Georgescu 12 Mar '08

12 Mar '08

Hello, There is a new Soap-Simple-Proxy release available. soap-simple-proxy (1.0.3) unstable; urgency=low * Allow items in SOAP policy object to be nillable * Added version replace string to WSDL file The software can be downloaded from: http://openxcap.org Kind regards, Adrian Georgescu

1 0

[OpenSER-Users] Increase MAX_BRANCHES value?
by Marcello Lupo 12 Mar '08

12 Mar '08

Hi to all, due to a failover routing policy (serial forking) i need to increase the MAX_BRANCHES value to something like 35. Our gateways have limitation of 30 Channels ---> 1 IP . I have 32 of them available. Can this change have any heavy and dangerous effect on openser ?? Just to know if i will need to increase memory aveilable or something like this. I already use : /*used only if PKG_MALLOC is defined*/ #define PKG_MEM_POOL_SIZE 4*1024*1024 due to my eavy use of AVP. Let me know if i need to increase it more than this. Thanks to all, Bye, Marcello

2 2

2025

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

sr-users