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[Serusers] unable to register SIP users after 24 hours (1 day)
by Ranjeet Kumar 06 Mar '07

06 Mar '07

Hi, I am using ser 0.9.6-2 on ubuntu edgy (linux) OS. After one day, it is not registering any user what may be the cause of this problem. Is it(daemon) stops/sleeps after some time of inactivity? Thanks, Ranjeet The information contained in, or attached to, this e-mail, contains confidential information and is intended solely for the use of the individual or entity to whom they are addressed and is subject to legal privilege. If you have received this e-mail in error you should notify the sender immediately by reply e-mail, delete the message from your system and notify your system manager. Please do not copy it for any purpose, or disclose its contents to any other person. The views or opinions presented in this e-mail are solely those of the author and do not necessarily represent those of the company. The recipient should check this e-mail and any attachments for the presence of viruses. The company accepts no liability for any damage caused, directly or indirectly, by any virus transmitted in this email. www.aztecsoft.com

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[Users] Subscriber table for GUI authentication
by alper ozbilen 06 Mar '07

06 Mar '07

Dear All, I am preparing php interface for web user authentication. But some fields on subscriber table confuse my mind. please let me write below : http://openser.org/dokuwiki/doku.php/database:1.2.x#subscriber 1st: How can we generate phplib_id value to keep phplib_id field? 2nd: I cant decide on username field whether I should generate username automatically or allow users to choose any available name or number ? 3th: what is the difference between username field and phone field at last: What is meaning of ha1 and ha1b field? any help wolud be appreciated Thanks a lot Alper Ozbilen _________________________________________________________________ Rates near 39yr lows! $430K Loan for $1,399/mo - Paying Too Much? Calculate new payment http://www.lowermybills.com/lre/index.jsp?sourceid=lmb-9632-18226&moid=7581

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[Serusers] ser 0.10.99 - unixsock_server.c:274: error: ‘last_process’ undeclared (first use in this function)
by "tzieleniewski" 06 Mar '07

06 Mar '07

Hi After a long way of trying to compile ser 0.10.99 I am still left some errors: Please help me with those: ser@tomix:~/src/ser-0.10.99$ make group_include="standard standard-dep stable" include_modules="xcap" all > ser.makeall.log make[1]: *** Nie podano obiektów i nie znaleziono makefile. Stop. make[1]: *** Nie podano obiektów i nie znaleziono makefile. Stop. make[1]: *** Nie podano obiektów i nie znaleziono makefile. Stop. unixsock_server.c: In function ‘init_unixsock_children’: unixsock_server.c:274: error: ‘last_process’ undeclared (first use in this function) unixsock_server.c:274: error: (Each undeclared identifier is reported only once unixsock_server.c:274: error: for each function it appears in.) unixsock_server.c:256: warning: unused variable ‘sockfd’ unixsock_server.c:253: warning: unused variable ‘backup’ make[1]: *** [unixsock_server.o] Błąd 1 checks.c: In function ‘generate_avps’: checks.c:89: error: ‘A_SER_ATTRS’ undeclared (first use in this function) checks.c:89: error: (Each undeclared identifier is reported only once checks.c:89: error: for each function it appears in.) checks.c: In function ‘radius_does_uri_exist’: checks.c:146: error: ‘V_CALL_CHECK’ undeclared (first use in this function) make[1]: *** [checks.o] Błąd 1 Errors in the above files correspond to the following lines: unixsock_server.c:274: error: snprintf(pt[last_process].desc, MAX_PT_DESC, "unix domain socket server @ %s", unixsock_name); checks.c:89: error: ‘A_SER_ATTRS’ undeclared (first use in this function) while ((vp = rc_avpair_get(vp, attrs[A_SER_ATTRS].v, 0))) { Bests Tomasz

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[Users] Dialog module bug?
by T.R. Missner 06 Mar '07

06 Mar '07

Hello, We are seeing this line in our logs quite consistently. Using dialog module with openser 1.1.1 I looked at the code and cannot determine why this condition would be happening. Any ideas? BUG:dialog:update_dlg_timer: links are nul (not in timer)! Thanks TR

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Re: [Serusers] Error Message
by Greger V. Teigre 06 Mar '07

06 Mar '07

Please copy the list. Strange, could you please post all the information about what you have done, your hardware configuration, your config, and the error trace to serdev? g-) Barton Fisher wrote: > Same results. I tried to compile after removing DSHM_MMAP from > Makefile.defs - no luck there either > That was only change I made after hello world test > > Bart > > Greger V. Teigre wrote: >> Hm, try a minimum (hello world) ser.cfg to see if you get the same >> error. PTR is not required. >> I assume you have not done any makefile changes. >> g-) >> >> Barton Fisher wrote: >>> Greger V. Teigre wrote: >>>> What is confusing? The message is pretty clear. But maybe the >>>> reason is unknown? >>>> What kind of system is this? Hardware? How much memory, etc etc? >>>> g-) >>>> >>>> >>>> Barton Fisher wrote: >>>>> A bit confused by startup message: >>>>> >>>>> 0(16102) ERROR: shm_mem_init: could not attach shared memory >>>>> segment: No space left on device >>>>> 0(16102) could not initialize shared memory pool, exiting... Too >>>>> much shared memory demanded: 33554432 >>>>> >>>>> I found prior message to upgrade Kernel to 2.4 - currently I'm on >>>>> 2.6.8-022stab076.1-enterprise >>>>> At another place it said to recompile ser with SYSV shm instead of >>>>> mmap (remove -DSHM_MMAP from Makefile.defs) <= This I'm not sure >>>>> exactly how / what to change in Makefile.def. >>>>> And yet another regarding no PTR <= I've contracted ISP regarding >>>>> this, but was hoping to bypass - I was hoping there was some >>>>> temporary fix for this until they act? >>>>> >>>>> I'm running SER 0.9.6 >>>>> >>>>> Thanks for any assistance. >>>>> >>>>> Bart >>>>> >>>>> >>>>> _______________________________________________ >>>>> Serusers mailing list >>>>> Serusers(a)lists.iptel.org >>>>> http://lists.iptel.org/mailman/listinfo/serusers >>> I have never seen PC, but here's what I know about it: >>> Processors = 4 >>> Model = Intel(R) XEON(TM) CPU 1.80GHz >>> Speed = 1.8GHz >>> Kernel Version = 2.6.8-022stab076.1-enterprise (SMP) >>> Distro Name = CentOS release 4.4 (Final) >>> Physical Memory = 3.89G >>> Me thinks issue is there is no PTR record as I see something like >>> this before it errors out: >>> >>> ... lots of junk ahead of this.... >>> 0(13373) if ( 0(13373) NOT( 0(13373) OR( 0(13373) uri 0(13373) =~ >>> 0(13373) "1 >>> 83" 0(13373) , 0(13373) uri 0(13373) =~ 0(13373) "200" 0(13373) ) >>> 0(13373) ) >>> 0(13373) ) { 0(13373) drop( 0(13373) type<0> 0(13373) ); 0(13373) } >>> else { 0(13 >>> 373) }; 0(13373) external_module_call( 0(13373) f_ptr<0x40573860> >>> 0(13373) , " >>> FA" 0(13373) ); 0(13373) external_module_call( 0(13373) >>> f_ptr<0x40573860> 0(13 >>> 373) , "FA" 0(13373) ); 0(13373) >>> >>> If it is PTR record checking, can I disable this checking? >>> Bart >>> >>> >>> >> >> >> >> __________ NOD32 2096 (20070305) Information __________ >> >> This message was checked by NOD32 antivirus system. >> http://www.eset.com >> >> >> > > > >

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[Users] Param in SIP header
by Michel Bensoussan 06 Mar '07

06 Mar '07

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[Serusers] re: npid=yes
by Alex Fler 05 Mar '07

05 Mar '07

Hello guys, Does anybody knows how to remove ";npdi=yes" from INVITE. INVITE sip:XXXXXXXXXXXXX;npdi=yes@server ip:5060;dtg=SIP;user=phone I've hear that subst_uri ,can be used . Can someone give me an example Thanks Alex Fler _____ I've stopped 5,370 spam and fraud messages. You can too! Free trial of spam and fraud protection at <http://www.cloudmark.com/sigs?rc=mrcr6> www.cloudmark.com <http://www.cloudmark.com/sigs?rc=mrcr6> Cloudmark Desktop - Join the fight against spam!

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Re: [Serusers] Error Message
by Greger V. Teigre 05 Mar '07

05 Mar '07

Hm, try a minimum (hello world) ser.cfg to see if you get the same error. PTR is not required. I assume you have not done any makefile changes. g-) Barton Fisher wrote: > Greger V. Teigre wrote: >> What is confusing? The message is pretty clear. But maybe the reason >> is unknown? >> What kind of system is this? Hardware? How much memory, etc etc? >> g-) >> >> >> Barton Fisher wrote: >>> A bit confused by startup message: >>> >>> 0(16102) ERROR: shm_mem_init: could not attach shared memory >>> segment: No space left on device >>> 0(16102) could not initialize shared memory pool, exiting... Too >>> much shared memory demanded: 33554432 >>> >>> I found prior message to upgrade Kernel to 2.4 - currently I'm on >>> 2.6.8-022stab076.1-enterprise >>> At another place it said to recompile ser with SYSV shm instead of >>> mmap (remove -DSHM_MMAP from Makefile.defs) <= This I'm not sure >>> exactly how / what to change in Makefile.def. >>> And yet another regarding no PTR <= I've contracted ISP regarding >>> this, but was hoping to bypass - I was hoping there was some >>> temporary fix for this until they act? >>> >>> I'm running SER 0.9.6 >>> >>> Thanks for any assistance. >>> >>> Bart >>> >>> >>> _______________________________________________ >>> Serusers mailing list >>> Serusers(a)lists.iptel.org >>> http://lists.iptel.org/mailman/listinfo/serusers > I have never seen PC, but here's what I know about it: > Processors = 4 > Model = Intel(R) XEON(TM) CPU 1.80GHz > Speed = 1.8GHz > Kernel Version = 2.6.8-022stab076.1-enterprise (SMP) > Distro Name = CentOS release 4.4 (Final) > Physical Memory = 3.89G > Me thinks issue is there is no PTR record as I see something like this > before it errors out: > > ... lots of junk ahead of this.... > 0(13373) if ( 0(13373) NOT( 0(13373) OR( 0(13373) uri 0(13373) =~ > 0(13373) "1 > 83" 0(13373) , 0(13373) uri 0(13373) =~ 0(13373) "200" 0(13373) ) > 0(13373) ) > 0(13373) ) { 0(13373) drop( 0(13373) type<0> 0(13373) ); 0(13373) } > else { 0(13 > 373) }; 0(13373) external_module_call( 0(13373) f_ptr<0x40573860> > 0(13373) , " > FA" 0(13373) ); 0(13373) external_module_call( 0(13373) > f_ptr<0x40573860> 0(13 > 373) , "FA" 0(13373) ); 0(13373) > > If it is PTR record checking, can I disable this checking? > Bart > > >

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[Serusers] 4 Rings and Fast Busy
by Ing. Marco Moran 05 Mar '07

05 Mar '07

Hi, I am having an issue when I call or someone calls among the registered users, SER only gives 4 rings to answer before it gives a fast-busy. Any suggestions? Thanks, Marco -----Mensaje original----- De: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] En nombre de serusers-request(a)lists.iptel.org Enviado el: Monday, March 05, 2007 3:40 AM Para: serusers(a)lists.iptel.org Asunto: Serusers Digest, Vol 47, Issue 5 Send Serusers mailing list submissions to serusers(a)lists.iptel.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.iptel.org/mailman/listinfo/serusers or, via email, send a message with subject or body 'help' to serusers-request(a)lists.iptel.org You can reach the person managing the list at serusers-owner(a)lists.iptel.org When replying, please edit your Subject line so it is more specific than "Re: Contents of Serusers digest..." Today's Topics: 1. Re: SIP SER (Greger V. Teigre) 2. Re: ask for ser performance parameters (Greger V. Teigre) 3. Re: test run (Greger V. Teigre) 4. Re: PATH Module in SER (0.9.6 or 0.10.x) (Greger V. Teigre) 5. Re: Problems creating DB in SER Ottendorf (Greger V. Teigre) 6. Re: speeddial - CANCEL MESSAGE ERROR (Greger V. Teigre) 7. Re: Error compiling eval module (Greger V. Teigre) 8. Re: Error Message (Greger V. Teigre) 9. RE: SIP SER (Goh, Tee-Yong) ---------------------------------------------------------------------- Message: 1 Date: Mon, 05 Mar 2007 08:02:52 +0100 From: "Greger V. Teigre" <greger(a)teigre.com> Subject: Re: [Serusers] SIP SER To: "Goh, Tee-Yong" <gohty(a)m1.com.sg> Cc: serusers(a)lists.iptel.org Message-ID: <45EBC09C.7050800(a)teigre.com> Content-Type: text/plain; charset="iso-8859-1" You can do that using exec command from ser.cfg when you receive a REGISTER. Use sipp or the internal SER stack to create new messages (see faqs on iptel.org) Or http://iptel.org/sems if you want to do it in a better way. g-) Goh, Tee-Yong wrote: > Hi Karim, > > Thank you for your advice. Noted your advice. > > Yes. My objective here is once the SER receives a REGISTER message > from one of the client, it will auto generate "INVITE" messages to the > clients. > > I am using the ser-0.9.6. I have looked through the coding as such > > /ser-0.9.6/modules/nathelper/nathelper.c > /ser-0.9.6/modules/nathelper/nathelper.cfg > /ser-0.9.6/modules/nathelper/nathelper_rrtpp.cfg > /ser-0.9.6/modules/registrar/save.c > /ser-0.9.6/modules/registrar/reply.c > /ser-0.9.6/modules/registrar/sip_msg.c > > Anyone knows where can i find the portion where once the SER receives > the REGISTER message, it responds with 200 OK to the client? Can i > find them inside nathelper or registrar or paser? > > Apologies if any inconvenience caused. > > Thank you. > > Regards, > Tee Yong > > > ------------------------------------------------------------------------ > *From:* karim basraoui [mailto:basraouik@gmail.com] > *Sent:* Sat 3/3/2007 8:36 PM > *To:* Goh, Tee-Yong > *Cc:* serusers(a)lists.iptel.org > *Subject:* Re: [Serusers] SIP SER > > Well, first of all a SER server does not generate messages it only > interprets the messages it receives. so it can not generate INVITE > messages. so if you want to call several clients from one client, this > one has to initiate the call by sending an INVITE to SER. > > When receiving the INVITE message, SER will forward it to the callees. > i did not try this yet, but i think that you can do this with the > method : "forward" as follows: > > if(method == INVITE) { > forward("callee1_IP_address","callee1_port"); > forward("callee2_IP_address","callee2_port"); > forward("callee3_IP_address","callee3_port"); > #and so on ... > } > > try this for example... and let me know if it works or not. > > ps: (personnally i think it won't work :p) > > 2007/3/3, Goh, Tee-Yong < gohty(a)m1.com.sg <mailto:gohty@m1.com.sg>>: > > Hi Karim, > > Yes you are right. > Please advice. > > Thank you. > > Regards, > Howard > > -----Original Message----- > *From:* karim basraoui [mailto:basraouik@gmail.com > <mailto:basraouik@gmail.com>] > *Sent:* Saturday, March 03, 2007 8:01 PM > *To:* Goh, Tee-Yong > *Cc:* serusers(a)lists.iptel.org <mailto:serusers@lists.iptel.org> > *Subject:* Re: [Serusers] SIP SER > > So if i understand, when a clint registers, SER will > automatically send invites to the other clients and make > connections with them? > > 2007/3/3, Goh, Tee-Yong < gohty(a)m1.com.sg > <mailto:gohty@m1.com.sg>>: > > Dear Karim, > > Thank you for your reply. > > Yes you are right. I have having difficulty searching for > this for past 2 months. > > Yes. Here is what i want. Once i receive the REGISTER > messages, my SER will send out "INVITE" message to my > other clients. This is what i am intended to do. However, > i could not search for the right coding inside the ser-0.9.6. > > Please advice and apologies if any inconvenience caused > for you. > > Thanks. > > Regards, > Howard > > -----Original Message----- > *From:* karim basraoui [mailto:basraouik@gmail.com > <mailto:basraouik@gmail.com>] > *Sent:* Saturday, March 03, 2007 7:18 PM > *To:* Goh, Tee-Yong > *Subject:* Re: [Serusers] SIP SER > > Hi, > > Do you want to make changes on SER behaviour when it > receives REGISTER messages. If it's the case you can > make a condition in the "ser.cfg" file, in the bloc > "route{ ... }". for example: > route { > if (method == "REGISTER") > { #what you want to do}; > } > > > 2007/3/3, Goh, Tee-Yong <gohty(a)m1.com.sg > <mailto:gohty@m1.com.sg>>: > > Hi, > > May i know from the SER-0.9.6, where can i look > for and make changes for the SER when receiving > "REGISTER" message and responsing " 200 OK" back > to the corresponding node or clients? > > Would appreciate your assistance and advice as > soon as possible as i have difficultly locating > the coding. Please advice where i can find the > script to edit the above mentioned messages. > > Apologies for the inconvenience caused. > > Thank you. > > Regards, > Howard > > ------------------------------------------------------------------------ > This email (including any attachment) is subject > to the following disclaimer: > http://m1.com.sg/M1/misc/disclaimer > <http://m1.com.sg/M1/misc/disclaimer> > ------------------------------------------------------------------------ > > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > <mailto:Serusers@lists.iptel.org> > http://lists.iptel.org/mailman/listinfo/serusers > > > > > -- > BASRAOUI Karim > www.basraouik.afrikart.net > <http://www.basraouik.afrikart.net> > > ------------------------------------------------------------------------ > This email (including any attachment) is subject to the > following disclaimer: http://m1.com.sg/M1/misc/disclaimer > <http://m1.com.sg/M1/misc/disclaimer> > ------------------------------------------------------------------------ > > > > > -- > BASRAOUI Karim > www.basraouik.afrikart.net <http://www.basraouik.afrikart.net> > > ------------------------------------------------------------------------ > This email (including any attachment) is subject to the following > disclaimer: http://m1.com.sg/M1/misc/disclaimer > ------------------------------------------------------------------------ > > > > > -- > BASRAOUI Karim > www.basraouik.afrikart.net <http://www.basraouik.afrikart.net> > ------------------------------------------------------------------------ > This email (including any attachment) is subject to the following > disclaimer: http://m1.com.sg/M1/misc/disclaimer > ------------------------------------------------------------------------ > ------------------------------------------------------------------------ > > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers >

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[Users] avp_check and AVP's with index's?
by Brandon Armstead 05 Mar '07

05 Mar '07

Hello, Can I simply do something like so? if(avp_check("$avp(i:1[1])", "eq/s:5/g")) { # true } else { # false } ?? I seem to get an error when I try to check the value against an AVP with index's, is there an easier way then copying the index's value into a full AVP, can't I just use the index's like normal AVP's? Thanks, all help is appreciated! -- Brandon Armstead

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Re: [Serusers] PATH Module in SER (0.9.6 or 0.10.x)
by SER LIST 05 Mar '07

05 Mar '07

How do I download the early path module for 0.9? Where did you post the patch? Thanks much in advance. >From: "Greger V. Teigre" <greger(a)teigre.com> >To: SER LIST <sergrp(a)hotmail.com> >CC: serusers(a)iptel.org >Subject: Re: [Serusers] PATH Module in SER (0.9.6 or 0.10.x) >Date: Mon, 05 Mar 2007 08:16:41 +0100 > >There's an early path module in experimental for 0.9. Also, a patch was >posted to serdev by Andreas Granig. For 0.10, AFAIK there is no path >support. I assume the patch can be ported without too much work. >I have reposted the patch and asked about status on serdev. >g-) > >SER LIST wrote: >>Is there a paths module for SER? I want to implement a Load balancer using >>Dispatcher module and I will need a path module for handling >>registrations. >>Thanks in advance >> >>_________________________________________________________________ >>http://local.live.com/default.aspx?v=2&cp=43.658648~-79.383962&style=r&lvl=… >> >> >>_______________________________________________ >>Serusers mailing list >>Serusers(a)lists.iptel.org >>http://lists.iptel.org/mailman/listinfo/serusers >> >> _________________________________________________________________ Free Alerts : Be smart - let your information find you ! http://alerts.live.com/Alerts/Default.aspx

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[Users] WeSIP part 2
by Helmut Kuper 05 Mar '07

05 Mar '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, I was too blind to see that binaries resp. startup scripts ... :( When I'm starting WeSIP I get this output: - ------------------------------------------------- hk@vm-en-test0:~/wesip/wesip/bin$ ./startup.sh Starting WESIP Application Server VozTelecom... hk@vm-en-test0:~/wesip/wesip/bin$ HttpConnector [main] - The HttpConnector [all:8080] has been created SipConnector [main] - The SipConnector [localhost] can not be launched. Revise the configuration. SipConnector [main] - javax.sip.PeerUnavailableException: The Peer SIP Stack: com.voztele.javax.sip.SipStackImpl could not be instantiated. Ensure the Path Name has been set. javax.sip.PeerUnavailableException: The Peer SIP Stack: com.voztele.javax.sip.SipStackImpl could not be instantiated. Ensure the Path Name has been set. at javax.sip.SipFactory.createStack(SipFactory.java:308) at javax.sip.SipFactory.createSipStack(SipFactory.java:147) at com.voztele.sipservlet.connector.SipConnector.start(SipConnector.java:238) at org.apache.catalina.core.StandardService.start(StandardService.java:454) at org.apache.catalina.core.StandardServer.start(StandardServer.java:2144) at org.apache.catalina.startup.Catalina.start(Catalina.java:471) at org.apache.catalina.startup.Catalina.execute(Catalina.java:358) at org.apache.catalina.startup.Catalina.process(Catalina.java:129) at sun.reflect.NativeMethodAccessorImpl.invoke0(Native Method) at sun.reflect.NativeMethodAccessorImpl.invoke(Unknown Source) at sun.reflect.DelegatingMethodAccessorImpl.invoke(Unknown Source) at java.lang.reflect.Method.invoke(Unknown Source) at org.apache.catalina.startup.Bootstrap.main(Bootstrap.java:156) Caused by: java.lang.reflect.InvocationTargetException at sun.reflect.NativeConstructorAccessorImpl.newInstance0(Native Method) at sun.reflect.NativeConstructorAccessorImpl.newInstance(Unknown Source) at sun.reflect.DelegatingConstructorAccessorImpl.newInstance(Unknown Source) at java.lang.reflect.Constructor.newInstance(Unknown Source) at javax.sip.SipFactory.createStack(SipFactory.java:302) ... 12 more Caused by: javax.sip.PeerUnavailableException: stack name is missing at com.voztele.javax.sip.SipStackImpl.<init>(SipStackImpl.java:292) ... 17 more after a few seconds the comes a second part: ConvergedHostDeployer [main] - Installing web application at context path from URL file:/home/hk/wesip/wesip/wesipapps/ROOT processJars: cannot find /WEB-INF/lib ConvergedContext [main] - ConvergedContext [Welcome] has been started StandardService [main] - > Default Wesip Application ==> http://localhost:8080/ > Manager Wesip Application ==> http://localhost:8080/manager/html? StandardService [main] - > In order to access the manager application use default credentials: - username: wesip - password: wesip > This configuration is defined in wesip-users.xml inside conf folder: - file: ./conf/wesip-users.xml - ----------------------------------------------------------------------- my server.cml config is <Server port="5005" shutdown="SHUTDOWN" debug="0"> <Service name="WeSIP-Standalone"> <Connector className="org.apache.catalina.connector.http.HttpConnector" port="8080" minProcessors="5" maxProcessors="75" enableLookups="true" acceptCount="10" debug="10" /> <Connector className="com.voztele.sipservlet.connector.SipConnector" minProcessors="5" maxProcessors="75" enableLookups="true" acceptCount="10" debug="10" addresses="localhost:5081" pathName="com.voztele" protocol="udp" > <ExtraProperties> <Property key="com.voztele.javax.sip.SER_ADDRESS" value="212.6.123.118" /> <Property key="com.voztele.javax.sip.SER_PORT" value="5081" /> <Property key="com.voztele.STATISTICS" value="false" /> <Property key="com.voztele.javax.sip.STACK_NAME" value="wesip_app_server_1" /> <Property key="com.voztele.javax.sip.THREAD_POOL_SIZE" value="100" /> <Property key="com.voztele.javax.sip.THREAD_POOL_SIZE" value="10" /> <Property key="com.voztele.javax.sip.REENTRANT_LISTENER" value="true" /> <Property key="com.voztele.javax.sip.AUTOMATIC_DIALOG_SUPPORT" value="false" /> <Property key="com.voztele.javax.sip.TRACE_LEVEL" value="0" /> </ExtraProperties> </Connector> <Engine name="Standalone" defaultHost="212.6.123.118" debug="1" className="com.voztele.sipservlet.core.ConvergedEngine"> <Mapper className="org.apache.catalina.core.StandardEngineMapper" protocol="HTTP/1.1"/> <Mapper className="org.apache.catalina.core.StandardEngineMapper" protocol="HTTP/1.0"/> <Mapper className="com.voztele.sipservlet.core.EngineSipMapper" protocol="SIP/2.0"/> <Logger className="org.apache.catalina.logger.FileLogger" timestamp="true"/> <Realm className="org.apache.catalina.realm.MemoryRealm" /> <Host name="localhost" nameSip="localhost_5081" debug="0" appBase="wesipapps" className="com.voztele.sipservlet.core.ConvergedHost"> <Logger className="org.apache.catalina.logger.FileLogger" timestamp="true"/> <Mapper className="com.voztele.sipservlet.core.SipHostMapper" protocol="SIP/2.0"/> <Mapper className="org.apache.catalina.core.HttpHostMapper" protocol="HTTP/1.1"/> <Mapper className="org.apache.catalina.core.HttpHostMapper" protocol="HTTP/1.0"/>  </Host> </Engine> </Service> </Server> - --------------------------------------------------------------------------- Have no idea how to solve this java execption. The missed "/WEB-INF/lib" isn't in /home/hk/wesip/wesip/wesipapps/ROOT but in /home/hk/wesip/wesip/server/webapps/manager ... Can I copy that directory ? Helmut - -- Mit freundlichen Grüßen Helmut Kuper Entwicklung Telefax: (0441) 8000-2799 mailto:helmut.kuper@ewetel.de ___________________________________ EWE TEL GmbH Cloppenburger Straße 310 26133 Oldenburg Handelsregister Amtsgericht Oldenburg HRB 3723 Vorsitzender des Aufsichtsrates: Heiko Harms Geschäftsführung: Hans-Joachim Iken, Dirk Thole, Jürgen Wehlend Homepage: http://www.ewetel.de ___________________________________ -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFF6+qR4tZeNddg3dwRAp1UAJsFkND1KaTByXA45mb8I2QkRgPjeACgv//R Y5kPDUongenLk3nZxmA2fRI= =NwKb -----END PGP SIGNATURE-----

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RE: [Serusers] SIP SER
by Goh, Tee-Yong 05 Mar '07

05 Mar '07

Dear Greger, Thank you for your advice. May i seek your advice where can i find the exec command from ser.cfg? Is there any reference? Thanks. Regards, Howard -----Original Message----- From: Greger V. Teigre [mailto:greger@teigre.com] Sent: Monday, March 05, 2007 3:03 PM To: Goh, Tee-Yong Cc: karim basraoui; serusers(a)lists.iptel.org Subject: Re: [Serusers] SIP SER You can do that using exec command from ser.cfg when you receive a REGISTER. Use sipp or the internal SER stack to create new messages (see faqs on iptel.org) Or http://iptel.org/sems if you want to do it in a better way. g-) Goh, Tee-Yong wrote: Hi Karim, Thank you for your advice. Noted your advice. Yes. My objective here is once the SER receives a REGISTER message from one of the client, it will auto generate "INVITE" messages to the clients. I am using the ser-0.9.6. I have looked through the coding as such /ser-0.9.6/modules/nathelper/nathelper.c /ser-0.9.6/modules/nathelper/nathelper.cfg /ser-0.9.6/modules/nathelper/nathelper_rrtpp.cfg /ser-0.9.6/modules/registrar/save.c /ser-0.9.6/modules/registrar/reply.c /ser-0.9.6/modules/registrar/sip_msg.c Anyone knows where can i find the portion where once the SER receives the REGISTER message, it responds with 200 OK to the client? Can i find them inside nathelper or registrar or paser? Apologies if any inconvenience caused. Thank you. Regards, Tee Yong _____ From: karim basraoui [ mailto:basraouik@gmail.com] Sent: Sat 3/3/2007 8:36 PM To: Goh, Tee-Yong Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] SIP SER Well, first of all a SER server does not generate messages it only interprets the messages it receives. so it can not generate INVITE messages. so if you want to call several clients from one client, this one has to initiate the call by sending an INVITE to SER. When receiving the INVITE message, SER will forward it to the callees. i did not try this yet, but i think that you can do this with the method : "forward" as follows: if(method == INVITE) { forward("callee1_IP_address","callee1_port"); forward("callee2_IP_address","callee2_port"); forward("callee3_IP_address","callee3_port"); #and so on ... } try this for example... and let me know if it works or not. ps: (personnally i think it won't work :p) 2007/3/3, Goh, Tee-Yong < gohty(a)m1.com.sg>: Hi Karim, Yes you are right. Please advice. Thank you. Regards, Howard -----Original Message----- From: karim basraoui [mailto: basraouik(a)gmail.com] Sent: Saturday, March 03, 2007 8:01 PM To: Goh, Tee-Yong Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] SIP SER So if i understand, when a clint registers, SER will automatically send invites to the other clients and make connections with them? 2007/3/3, Goh, Tee-Yong < <mailto:gohty@m1.com.sg> gohty(a)m1.com.sg>: Dear Karim, Thank you for your reply. Yes you are right. I have having difficulty searching for this for past 2 months. Yes. Here is what i want. Once i receive the REGISTER messages, my SER will send out "INVITE" message to my other clients. This is what i am intended to do. However, i could not search for the right coding inside the ser-0.9.6. Please advice and apologies if any inconvenience caused for you. Thanks. Regards, Howard -----Original Message----- From: karim basraoui [mailto: basraouik(a)gmail.com] Sent: Saturday, March 03, 2007 7:18 PM To: Goh, Tee-Yong Subject: Re: [Serusers] SIP SER Hi, Do you want to make changes on SER behaviour when it receives REGISTER messages. If it's the case you can make a condition in the "ser.cfg" file, in the bloc "route{ ... }". for example: route { if (method == "REGISTER") { #what you want to do}; } 2007/3/3, Goh, Tee-Yong < gohty(a)m1.com.sg <mailto:gohty@m1.com.sg> >: Hi, May i know from the SER-0.9.6, where can i look for and make changes for the SER when receiving "REGISTER" message and responsing " 200 OK" back to the corresponding node or clients? Would appreciate your assistance and advice as soon as possible as i have difficultly locating the coding. Please advice where i can find the script to edit the above mentioned messages. Apologies for the inconvenience caused. Thank you. Regards, Howard _____ This email (including any attachment) is subject to the following disclaimer: http://m1.com.sg/M1/misc/disclaimer <http://m1.com.sg/M1/misc/disclaimer> _____ _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers -- BASRAOUI Karim www.basraouik.afrikart.net _____ This email (including any attachment) is subject to the following disclaimer: http://m1.com.sg/M1/misc/disclaimer <http://m1.com.sg/M1/misc/disclaimer> _____ -- BASRAOUI Karim www.basraouik.afrikart.net _____ This email (including any attachment) is subject to the following disclaimer: http://m1.com.sg/M1/misc/disclaimer _____ -- BASRAOUI Karim www.basraouik.afrikart.net _____ This email (including any attachment) is subject to the following disclaimer: http://m1.com.sg/M1/misc/disclaimer _____ _____ _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers This email (including any attachment) is subject to the following disclaimer: http://m1.com.sg/M1/misc/disclaimer

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[Serusers] Error Message
by Barton Fisher 05 Mar '07

05 Mar '07

A bit confused by startup message: 0(16102) ERROR: shm_mem_init: could not attach shared memory segment: No space left on device 0(16102) could not initialize shared memory pool, exiting... Too much shared memory demanded: 33554432 I found prior message to upgrade Kernel to 2.4 - currently I'm on 2.6.8-022stab076.1-enterprise At another place it said to recompile ser with SYSV shm instead of mmap (remove -DSHM_MMAP from Makefile.defs) <= This I'm not sure exactly how / what to change in Makefile.def. And yet another regarding no PTR <= I've contracted ISP regarding this, but was hoping to bypass - I was hoping there was some temporary fix for this until they act? I'm running SER 0.9.6 Thanks for any assistance. Bart

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[Serusers] Error compiling eval module
by Andrey Kuprianov 05 Mar '07

05 Mar '07

Guys, I've just updated sip_router from CVS. Cant compile eval module no more. It says, eval.c:33:23: error: uuid/uuid.h: No such file or directory eval.c: In function 'sel_gen_uuid': eval.c:1975: error: 'uuid_t' undeclared (first use in this function) eval.c:1975: error: (Each undeclared identifier is reported only once eval.c:1975: error: for each function it appears in.) eval.c:1975: error: expected ';' before 'u' eval.c:1976: warning: implicit declaration of function 'uuid_generate' eval.c:1976: error: 'u' undeclared (first use in this function) eval.c:1977: warning: implicit declaration of function 'uuid_unparse' make[1]: *** [eval.o] Error 1 make[1]: Leaving directory `/home/andrey/sip_router/modules/eval' make: *** [modules] Error 1 Plz, help :)) Andrey.

3 3

[Serusers] speeddial - CANCEL MESSAGE ERROR
by tzieleniewski 05 Mar '07

05 Mar '07

Hi!! I was utilizing the speeddial functionality and I encountered the following error: When user sends a cancel message before the destination leg answers the call the cancel message is not properly routed, because sd_lookup("speed_dial") returns negative result for the processing of cancel message. If other side answers everything is ok (BYE etc). here is my Speed_dial logic: if (uri =~ "sip:[0-9]{2}@.*") { if(sd_lookup("speed_dial")) { route(FORWARD); drop; } } Bests tomasz

2 1

[Serusers] Problems creating DB in SER Ottendorf
by Andrey Kuprianov 05 Mar '07

05 Mar '07

Hi, Im running Ubuntu with MySQL 5.0.24a. I've downloaded and compiled Ottendorf from CVS successfully. The problem came, when I tried to create db with sip_router/scripts/mysql/ser_mysql.sh create. It asked for root password (i dont have any so I pressed Enter right away), but then displayed mysql options (give below) for me instead of creating DB. I suppose some argument went missing. How can I fix it? I also remember there was a manual way to do it, but I couldnt find a proper link. Can anyone help me, plz? Output: mysql Ver 14.12 Distrib 5.0.24a, for pc-linux-gnu (i486) using readline 5.1 Copyright (C) 2002 MySQL AB This software comes with ABSOLUTELY NO WARRANTY. This is free software, and you are welcome to modify and redistribute it under the GPL license Usage: mysql [OPTIONS] [database] -?, --help Display this help and exit. -I, --help Synonym for -? --auto-rehash Enable automatic rehashing. One doesn't need to use 'rehash' to get table and field completion, but startup and reconnecting may take a longer time. Disable with --disable-auto-rehash. -A, --no-auto-rehash No automatic rehashing. One has to use 'rehash' to get table and field completion. This gives a quicker start of mysql and disables rehashing on reconnect. WARNING: options deprecated; use --disable-auto-rehash instead. -B, --batch Don't use history file. Disable interactive behavior. (Enables --silent) --character-sets-dir=name Directory where character sets are. --default-character-set=name Set the default character set. -C, --compress Use compression in server/client protocol. -#, --debug[=#] This is a non-debug version. Catch this and exit -D, --database=name Database to use. --delimiter=name Delimiter to be used. -e, --execute=name Execute command and quit. (Disables --force and history file) -E, --vertical Print the output of a query (rows) vertically. -f, --force Continue even if we get an sql error. -g, --no-named-commands Named commands are disabled. Use \* form only, or use named commands only in the beginning of a line ending with a semicolon (;) Since version 10.9 the client now starts with this option ENABLED by default! Disable with '-G'. Long format commands still work from the first line. WARNING: option deprecated; use --disable-named-commands instead. -G, --named-commands Enable named commands. Named commands mean this program's internal commands; see mysql> help . When enabled, the named commands can be used from any line of the query, otherwise only from the first line, before an enter. Disable with --disable-named-commands. This option is disabled by default. -i, --ignore-spaces Ignore space after function names. --local-infile Enable/disable LOAD DATA LOCAL INFILE. -b, --no-beep Turn off beep on error. -h, --host=name Connect to host. -H, --html Produce HTML output. -X, --xml Produce XML output --line-numbers Write line numbers for errors. -L, --skip-line-numbers Don't write line number for errors. WARNING: -L is deprecated, use long version of this option instead. --no-pager Disable pager and print to stdout. See interactive help (\h) also. WARNING: option deprecated; use --disable-pager instead. --no-tee Disable outfile. See interactive help (\h) also. WARNING: option deprecated; use --disable-tee instead -n, --unbuffered Flush buffer after each query. --column-names Write column names in results. -N, --skip-column-names Don't write column names in results. WARNING: -N is deprecated, use long version of this options instead. -O, --set-variable=name Change the value of a variable. Please note that this option is deprecated; you can set variables directly with --variable-name=value. --sigint-ignore Ignore SIGINT (CTRL-C) -o, --one-database Only update the default database. This is useful for skipping updates to other database in the update log. --pager[=name] Pager to use to display results. If you don't supply an option the default pager is taken from your ENV variable PAGER. Valid pagers are less, more, cat [> filename], etc. See interactive help (\h) also. This option does not work in batch mode. -p, --password[=name] Password to use when connecting to server. If password is not given it's asked from the tty. WARNING: This is insecure as the password is visible for anyone through /proc for a short time. -P, --port=# Port number to use for connection. --prompt=name Set the mysql prompt to this value. --protocol=name The protocol of connection (tcp,socket,pipe,memory). -q, --quick Don't cache result, print it row by row. This may slow down the server if the output is suspended. Doesn't use history file. -r, --raw Write fields without conversion. Used with --batch. --reconnect Reconnect if the connection is lost. Disable with --disable-reconnect. This option is enabled by default. -s, --silent Be more silent. Print results with a tab as separator, each row on new line. -S, --socket=name Socket file to use for connection. --ssl Enable SSL for connection (automatically enabled with other flags). Disable with --skip-ssl. --ssl-ca=name CA file in PEM format (check OpenSSL docs, implies --ssl). --ssl-capath=name CA directory (check OpenSSL docs, implies --ssl). --ssl-cert=name X509 cert in PEM format (implies --ssl). --ssl-cipher=name SSL cipher to use (implies --ssl). --ssl-key=name X509 key in PEM format (implies --ssl). --ssl-verify-server-cert Verify server's "Common Name" in its cert against hostname used when connecting. This option is disabled by default. -t, --table Output in table format. -T, --debug-info Print some debug info at exit. --tee=name Append everything into outfile. See interactive help (\h) also. Does not work in batch mode. -u, --user=name User for login if not current user. -U, --safe-updates Only allow UPDATE and DELETE that uses keys. -U, --i-am-a-dummy Synonym for option --safe-updates, -U. -v, --verbose Write more. (-v -v -v gives the table output format). -V, --version Output version information and exit. -w, --wait Wait and retry if connection is down. --connect_timeout=# Number of seconds before connection timeout. --max_allowed_packet=# Max packet length to send to, or receive from server --net_buffer_length=# Buffer for TCP/IP and socket communication --select_limit=# Automatic limit for SELECT when using --safe-updates --max_join_size=# Automatic limit for rows in a join when using --safe-updates --secure-auth Refuse client connecting to server if it uses old (pre-4.1.1) protocol --show-warnings Show warnings after every statement. Default options are read from the following files in the given order: /etc/mysql/my.cnf ~/.my.cnf /etc/mysql/my.cnf The following groups are read: mysql client The following options may be given as the first argument: --print-defaults Print the program argument list and exit --no-defaults Don't read default options from any options file --defaults-file=# Only read default options from the given file # --defaults-extra-file=# Read this file after the global files are read Variables (--variable-name=value) and boolean options {FALSE|TRUE} Value (after reading options) --------------------------------- ----------------------------- auto-rehash TRUE character-sets-dir (No default value) default-character-set latin1 compress FALSE database (No default value) delimiter ; vertical FALSE force FALSE named-commands FALSE local-infile FALSE no-beep FALSE host localhost html FALSE xml FALSE line-numbers TRUE unbuffered FALSE column-names TRUE sigint-ignore FALSE port 3306 prompt mysql> quick FALSE raw FALSE reconnect FALSE socket /var/run/mysqld/mysqld.sock ssl FALSE ssl-ca (No default value) ssl-capath (No default value) ssl-cert (No default value) ssl-cipher (No default value) ssl-key (No default value) ssl-verify-server-cert FALSE table FALSE debug-info FALSE user root safe-updates FALSE i-am-a-dummy FALSE connect_timeout 0 max_allowed_packet 16777216 net_buffer_length 16384 select_limit 1000 max_join_size 1000000 secure-auth FALSE show-warnings FALSE Andrey.

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[Serusers] PATH Module in SER (0.9.6 or 0.10.x)
by SER LIST 05 Mar '07

05 Mar '07

Is there a paths module for SER? I want to implement a Load balancer using Dispatcher module and I will need a path module for handling registrations. Thanks in advance _________________________________________________________________ http://local.live.com/default.aspx?v=2&cp=43.658648~-79.383962&style=r&lvl=…

2 1

[Serusers] ask for ser performance parameters
by Su Kerry 05 Mar '07

05 Mar '07

3 3

[Users] WeSIP b2bua problem
by tele 04 Mar '07

04 Mar '07

Hi all, I've read on the WeSIP website that the support is made through openser mailing-list, so here i'am. I've problem with the SIP Servlet in b2bua mode, the problem is that the ACK is not forwarded to the other call-leg. INVITE, 183 session progress, 180 ringing are correctly forwarded but the ACK no. In the debug i can't see too much information about the request is not forwarded. Attached there is a .zip with all the log, trace and configuration. thanks for the support, :tele

1 0

Re: FW: [Users] imc
by Daniel-Constantin Mierla 03 Mar '07

03 Mar '07

Hello, because the registrar module imports now some values of usrloc parameters it must be loaded before the usrloc module. http://openser.org/dokuwiki/doku.php/install:1.1.x-to-1.2.x#registrar use: loadmodule "usrloc.so" loadmodule "registrar.so" Cheers, Daniel On 03/03/07 15:14, Henri Keski-Sikkilä wrote: > > Daniel-Constantin Mierla wrote: > > Hello, > > > > henri keski-sikkila wrote: > > > >> [...] > >> > >> When user exit room only admin gets message "The user [user] has left > >> the room" others doesn't know anything that user has left room... > > > > Could you test with the latest CVS version? If you get the same, please > > capture the sip traffic on openser box and send it to me for this case > > ngrep -d any -qt port 5060 > > > > Cheers, > > Daniel > > > > Hello Daniel, > > I took the latest source code from svn. I used a default routing script > "imc.cfg" that comes with the imc module. When I started openser, I got > these messages: > > 0(0) imc:mod_init:the query returned no result > 0(0) find_cmd_export_t: found <load_tm>(0) in module tm > [modules/tm/tm.so] > 0(0) pool_remove: Removing connection from the pool > 0(0) DEBUG: register_stats: imc > 0(0) DEBUG: register_mi: imc > 0(0) DEBUG: init_mod: tm > 0(0) TM - initializing... > 0(0) Call-ID initialization: '7fbb9de7' > 0(0) unixsock_register_cmd: New command (t_uac_dlg) registered > 0(0) unixsock_register_cmd: New command (t_uac_cancel) registered > 0(0) unixsock_register_cmd: New command (t_hash) registered > 0(0) unixsock_register_cmd: New command (t_reply) registered > 0(0) DEBUG: lock_initialize: lock initialization started > 0(0) DEBUG: MD5 calculated: 72dbedf8f6d9bc423ac07e79fbac2ddb > 0(0) DEBUG: MD5 calculated: ce1d8ecb1b6d4462277ac0abb13e18fc > 0(0) DEBUG: register_stats: tm > 0(0) DEBUG: register_mi: tm > 0(0) DEBUG: init_mod: xlog > 0(0) XLOG: initializing ... > 0(0) DEBUG: init_mod: registrar > 0(0) registrar - initializing > 0(0) find_cmd_export_t: found <load_sl>(0) in module sl > [modules/sl/sl.so] > 0(0) find_cmd_export_t: found <ul_bind_usrloc>(1) in module usrloc > [modules/usrloc/usrloc.so] > 0(0) ERROR:usrloc:bind_usrloc: configuration error - trying to bind to > usrlo module before being initialized > 0(0) init_mod(): Error while initializing module registrar > ERROR: error while initializing modules > 0(0) INFO:mi_fifo:mi_destroy:memory for the child's mi_fifo_pid was > not allocated -> nothing to destroy > 0(0) XLOG: destroy module ... > 0(0) DEBUG: tm_shutdown : start > 0(0) DEBUG: unlink_timer_lists : emptying DELETE list > 0(0) DEBUG: tm_shutdown : emptying hash table > 0(0) DEBUG: tm_shutdown : releasing timers > 0(0) DEBUG: tm_shutdown : removing semaphores > 0(0) DEBUG: tm_shutdown : destroying tmcb lists > 0(0) DEBUG: tm_shutdown : done > 0(0) imc: destroy module ... > 0(15800) shm_mem_destroy > 0(15800) destroying the shared memory lock > > > Best Regards, > Henri > >

1 0

Re: [Serusers] SIP SER
by karim basraoui 03 Mar '07

03 Mar '07

So if i understand, when a clint registers, SER will automatically send invites to the other clients and make connections with them? 2007/3/3, Goh, Tee-Yong <gohty(a)m1.com.sg>: > > Dear Karim, > > Thank you for your reply. > > Yes you are right. I have having difficulty searching for this for past 2 > months. > > Yes. Here is what i want. Once i receive the REGISTER messages, my SER > will send out "INVITE" message to my other clients. This is what i am > intended to do. However, i could not search for the right coding inside the > ser-0.9.6. > > Please advice and apologies if any inconvenience caused for you. > > Thanks. > > Regards, > Howard > > -----Original Message----- > *From:* karim basraoui [mailto:basraouik@gmail.com] > *Sent:* Saturday, March 03, 2007 7:18 PM > *To:* Goh, Tee-Yong > *Subject:* Re: [Serusers] SIP SER > > Hi, > > Do you want to make changes on SER behaviour when it receives REGISTER > messages. If it's the case you can make a condition in the "ser.cfg" file, > in the bloc "route{ ... }". for example: > route { > if (method == "REGISTER") > { #what you want to do}; > } > > > 2007/3/3, Goh, Tee-Yong <gohty(a)m1.com.sg >: > > > > Hi, > > > > May i know from the SER-0.9.6, where can i look for and make changes for > > the SER when receiving "REGISTER" message and responsing " 200 OK" back to > > the corresponding node or clients? > > > > Would appreciate your assistance and advice as soon as possible as i > > have difficultly locating the coding. Please advice where i can find the > > script to edit the above mentioned messages. > > > > Apologies for the inconvenience caused. > > > > Thank you. > > > > Regards, > > Howard > > > > ------------------------------ > > This email (including any attachment) is subject to the following > > disclaimer: http://m1.com.sg/M1/misc/disclaimer > > ------------------------------ > > > > _______________________________________________ > > Serusers mailing list > > Serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > > > -- > BASRAOUI Karim > www.basraouik.afrikart.net > > ------------------------------ > This email (including any attachment) is subject to the following > disclaimer: http://m1.com.sg/M1/misc/disclaimer > ------------------------------ > -- BASRAOUI Karim www.basraouik.afrikart.net

1 0

[Serusers] SIP SER
by Goh, Tee-Yong 03 Mar '07

03 Mar '07

Hi, May i know from the SER-0.9.6, where can i look for and make changes for the SER when receiving "REGISTER" message and responsing " 200 OK" back to the corresponding node or clients? Would appreciate your assistance and advice as soon as possible as i have difficultly locating the coding. Please advice where i can find the script to edit the above mentioned messages. Apologies for the inconvenience caused. Thank you. Regards, Howard This email (including any attachment) is subject to the following disclaimer: http://m1.com.sg/M1/misc/disclaimer

1 0

[Users] regarding SIPS/TLS
by Andrew T Gin 02 Mar '07

02 Mar '07

Hi I'm not sure whether this belongs in the developer one or not. When I proxy requests through openSER, and the contact address for the client uses a secure URI (SIPS) and the protocol stated is TLS, I get the following error: 7(6639) ERROR: uri2proxy: bad transport for sips uri: 3 7(6639) ERROR:tm:t_forward_nonack: failure to add branches This is in ut.h, line 113. If the URI is a secure one (SIPS), why must the protocol be TCP or NONE? openSER exits when the URI is secure (SIPS) AND the transport is TLS. Why is this?

2 1

[Users] Module UAC to authentication and redirect calls
by Thiago Maluf 02 Mar '07

02 Mar '07

Hi List, I need config one module UAC to do authentication and redirect my calls to other Voip Provider. Somebody already did it? If can tell me what functions you used. Thanks in advanced! Thiago Maluf. ---------------------------------------------------------------- THIAGO MALUF RESENDE Consultor Voip e Programador WEB (Voip Developer and Web Developer) e-mail: malufrj(a)gmail.com

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[Users] OpenSer 1.1 crashs on TLS
by Ohad.Levy＠infineon.com 02 Mar '07

02 Mar '07

Hi, I've used the default configuration for TLS, and tried to register one snom 320 phone, openser crashes immediately. Attached is a log - any ideas? Thanks, Ohad 19(14449) tcpconn_new: new tcp connection to: 172.28.129.126 19(14449) tcpconn_new: on port 2358, type 3 19(14449) tls_tcpconn_init: Entered: Creating a whole new ssl connection 19(14449) tls_tcpconn_init: Looking up socket based TLS server domain [172.29.162.26:5061] 19(14449) tls_find_server_domain: virtual TLS server domain not found, Using default TLS server domain settings 19(14449) tls_tcpconn_init: Found socket based TLS server domain [0.0.0.0:0] 19(14449) tls_tcpconn_init: Setting in ACCEPT mode (server) 19(14449) tcpconn_add: hashes: 410, 1 19(14449) handle_new_connect: new connection: 0xb60b93f8 36 flags: 0002 19(14449) send2child: to tcp child 0 11(14441), 0xb60b93f8 11(14441) received n=4 con=0xb60b93f8, fd=23 11(14441) DBG: io_watch_add(0x80fea60, 23, 2, 0xb60b93f8), fd_no=1 11(14441) tls_update_fd: New fd is 23 19(14449) ERROR: receive_fd: EOF on 18 19(14449) DBG: handle_ser_child: dead child 11, pid 14441 (shutting down?) 19(14449) DBG: io_watch_del (0x80fe900, 18, -1, 0x0) fd_no=27 called 19(14449) DBG: handle_tcp_child: dead tcp child 0 (pid 14441, no 11) (shutting down?) 19(14449) DBG: io_watch_del (0x80fe900, 20, -1, 0x0) fd_no=26 called 0(14430) child process 14441 exited by a signal 11 0(14430) core was generated 0(14430) INFO: terminating due to SIGCHLD 1(14431) INFO: signal 15 received 1(14431) Memory status (pkg): 1(14431) fm_status (0x8130780): 1(14431) heap size= 1048576 1(14431) used= 22016, used+overhead=40752, free=1007824 1(14431) max used (+overhead)= 40752 1(14431) dumping free list: 1(14431) hash = 7 fragments no.: 1, unused: 0 bucket size: 56 - 56 (first 56) 1(14431) hash = 16 fragments no.: 1, unused: 0 bucket size: 128 - 128 (first 128) 1(14431) hash = 92 fragments no.: 1, unused: 0 bucket size: 736 - 736 (first 736) 1(14431) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 1006904) 2(14432) INFO: signal 15 received 2(14432) Memory status (pkg): 2(14432) fm_status (0x8130780): 2(14432) heap size= 1048576 2(14432) used= 21888, used+overhead=40576, free=1008000 2(14432) max used (+overhead)= 40608 2(14432) dumping free list: 2(14432) hash = 2 fragments no.: 1, unused: 0 bucket size: 16 - 16 (first 16) 2(14432) hash = 11 fragments no.: 1, unused: 0 bucket size: 88 - 88 (first 88) 2(14432) hash = 16 fragments no.: 2, unused: 0 bucket size: 128 - 128 (first 128) 2(14432) hash = 92 fragments no.: 1, unused: 0 bucket size: 736 - 736 (first 736) 2(14432) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 1006904) 2(14432) TOTAL: 6 free fragments = 1008000 free bytes 2(14432) ----------------------------- 3(14433) INFO: signal 15 received 3(14433) Memory status (pkg): 3(14433) fm_status (0x8130780): 3(14433) heap size= 1048576 3(14433) used= 21888, used+overhead=40576, free=1008000 3(14433) max used (+overhead)= 40608 3(14433) dumping free list: 3(14433) hash = 2 fragments no.: 1, unused: 0 bucket size: 16 - 16 (first 16) 3(14433) hash = 11 fragments no.: 1, unused: 0 bucket size: 88 - 88 (first 88) 3(14433) hash = 16 fragments no.: 2, unused: 0 bucket size: 128 - 128 (first 128) 3(14433) hash = 92 fragments no.: 1, unused: 0 bucket size: 736 - 736 (first 736) 3(14433) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 1006904) 3(14433) TOTAL: 6 free fragments = 1008000 free bytes 3(14433) ----------------------------- 4(14434) INFO: signal 15 received 4(14434) Memory status (pkg): 4(14434) fm_status (0x8130780): 4(14434) heap size= 1048576 4(14434) used= 21888, used+overhead=40576, free=1008000 4(14434) max used (+overhead)= 40608 4(14434) dumping free list: 4(14434) hash = 2 fragments no.: 1, unused: 0 bucket size: 16 - 16 (first 16) 4(14434) hash = 11 fragments no.: 1, unused: 0 bucket size: 88 - 88 (first 88) 4(14434) hash = 16 fragments no.: 2, unused: 0 bucket size: 128 - 128 (first 128) 4(14434) hash = 92 fragments no.: 1, unused: 0 bucket size: 736 - 736 (first 736) 4(14434) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 1006904) 4(14434) TOTAL: 6 free fragments = 1008000 free bytes 4(14434) ----------------------------- 5(14435) INFO: signal 15 received 5(14435) Memory status (pkg): 5(14435) fm_status (0x8130780): 5(14435) heap size= 1048576 5(14435) used= 21888, used+overhead=40576, free=1008000 5(14435) max used (+overhead)= 40608 5(14435) dumping free list: 5(14435) hash = 2 fragments no.: 1, unused: 0 bucket size: 16 - 16 (first 16) 5(14435) hash = 11 fragments no.: 1, unused: 0 bucket size: 88 - 88 (first 88) 5(14435) hash = 16 fragments no.: 2, unused: 0 bucket size: 128 - 128 (first 128) 5(14435) hash = 92 fragments no.: 1, unused: 0 bucket size: 736 - 736 (first 736) 5(14435) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 1006904) 5(14435) TOTAL: 6 free fragments = 1008000 free bytes 5(14435) ----------------------------- 6(14436) INFO: signal 15 received 6(14436) Memory status (pkg): 6(14436) fm_status (0x8130780): 6(14436) heap size= 1048576 6(14436) used= 21888, used+overhead=40576, free=1008000 6(14436) max used (+overhead)= 40608 6(14436) dumping free list: 6(14436) hash = 2 fragments no.: 1, unused: 0 bucket size: 16 - 16 (first 16) 6(14436) hash = 11 fragments no.: 1, unused: 0 bucket size: 88 - 88 (first 88) 6(14436) hash = 16 fragments no.: 2, unused: 0 bucket size: 128 - 128 (first 128) 6(14436) hash = 92 fragments no.: 1, unused: 0 bucket size: 736 - 736 (first 736) 6(14436) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 1006904) 6(14436) TOTAL: 6 free fragments = 1008000 free bytes 6(14436) ----------------------------- 7(14437) INFO: signal 15 received 7(14437) Memory status (pkg): 7(14437) fm_status (0x8130780): 7(14437) heap size= 1048576 7(14437) used= 21888, used+overhead=40576, free=1008000 7(14437) max used (+overhead)= 40608 7(14437) dumping free list: 7(14437) hash = 2 fragments no.: 1, unused: 0 bucket size: 16 - 16 (first 16) 7(14437) hash = 11 fragments no.: 1, unused: 0 bucket size: 88 - 88 (first 88) 7(14437) hash = 16 fragments no.: 2, unused: 0 bucket size: 128 - 128 (first 128) 7(14437) hash = 92 fragments no.: 1, unused: 0 bucket size: 736 - 736 (first 736) 7(14437) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 1006904) 7(14437) TOTAL: 6 free fragments = 1008000 free bytes 7(14437) ----------------------------- 8(14438) INFO: signal 15 received 8(14438) Memory status (pkg): 8(14438) fm_status (0x8130780): 8(14438) heap size= 1048576 8(14438) used= 21888, used+overhead=40576, free=1008000 8(14438) max used (+overhead)= 40608 8(14438) dumping free list: 8(14438) hash = 2 fragments no.: 1, unused: 0 bucket size: 16 - 16 (first 16) 8(14438) hash = 11 fragments no.: 1, unused: 0 bucket size: 88 - 88 (first 88) 8(14438) hash = 16 fragments no.: 2, unused: 0 bucket size: 128 - 128 (first 128) 8(14438) hash = 92 fragments no.: 1, unused: 0 bucket size: 736 - 736 (first 736) 8(14438) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 1006904) 8(14438) TOTAL: 6 free fragments = 1008000 free bytes 8(14438) ----------------------------- 9(14439) INFO: signal 15 received 9(14439) Memory status (pkg): 9(14439) fm_status (0x8130780): 9(14439) heap size= 1048576 9(14439) used= 21888, used+overhead=40576, free=1008000 9(14439) max used (+overhead)= 40608 9(14439) dumping free list: 9(14439) hash = 2 fragments no.: 1, unused: 0 bucket size: 16 - 16 (first 16) 9(14439) hash = 11 fragments no.: 1, unused: 0 bucket size: 88 - 88 (first 88) 9(14439) hash = 16 fragments no.: 2, unused: 0 bucket size: 128 - 128 (first 128) 9(14439) hash = 92 fragments no.: 1, unused: 0 bucket size: 736 - 736 (first 736) 9(14439) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 1006904) 9(14439) TOTAL: 6 free fragments = 1008000 free bytes 9(14439) ----------------------------- 10(14440) INFO: signal 15 received 10(14440) Memory status (pkg): 10(14440) fm_status (0x8130780): 10(14440) heap size= 1048576 10(14440) used= 21856, used+overhead=40536, free=1008040 10(14440) max used (+overhead)= 40608 10(14440) dumping free list: 10(14440) hash = 2 fragments no.: 1, unused: 0 bucket size: 16 - 16 (first 16) 10(14440) hash = 16 fragments no.: 3, unused: 0 bucket size: 128 - 128 (first 128) 10(14440) hash = 92 fragments no.: 1, unused: 0 bucket size: 736 - 736 (first 736) 10(14440) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 1006904) 10(14440) TOTAL: 6 free fragments = 1008040 free bytes 10(14440) ----------------------------- 12(14442) INFO: signal 15 received 12(14442) Memory status (pkg): 12(14442) fm_status (0x8130780): 12(14442) heap size= 1048576 12(14442) used= 72048, used+overhead=90744, free=957832 12(14442) max used (+overhead)= 90744 12(14442) dumping free list: 12(14442) hash = 2 fragments no.: 1, unused: 0 bucket size: 16 - 16 (first 16) 12(14442) hash = 16 fragments no.: 3, unused: 0 bucket size: 128 - 128 (first 128) 12(14442) hash = 92 fragments no.: 1, unused: 0 bucket size: 736 - 736 (first 736) 12(14442) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 956696) 12(14442) TOTAL: 6 free fragments = 957832 free bytes 12(14442) ----------------------------- 13(14443) INFO: signal 15 received 13(14443) Memory status (pkg): 13(14443) fm_status (0x8130780): 13(14443) heap size= 1048576 13(14443) used= 72048, used+overhead=90744, free=957832 13(14443) max used (+overhead)= 90744 13(14443) dumping free list: 13(14443) hash = 2 fragments no.: 1, unused: 0 bucket size: 16 - 16 (first 16) 13(14443) hash = 16 fragments no.: 3, unused: 0 bucket size: 128 - 128 (first 128) 13(14443) hash = 92 fragments no.: 1, unused: 0 bucket size: 736 - 736 (first 736) 13(14443) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 956696) 13(14443) TOTAL: 6 free fragments = 957832 free bytes 13(14443) ----------------------------- 14(14444) INFO: signal 15 received 14(14444) Memory status (pkg): 14(14444) fm_status (0x8130780): 14(14444) heap size= 1048576 14(14444) used= 72048, used+overhead=90744, free=957832 14(14444) max used (+overhead)= 90744 14(14444) dumping free list: 14(14444) hash = 2 fragments no.: 1, unused: 0 bucket size: 16 - 16 (first 16) 14(14444) hash = 16 fragments no.: 3, unused: 0 bucket size: 128 - 128 (first 128) 14(14444) hash = 92 fragments no.: 1, unused: 0 bucket size: 736 - 736 (first 736) 14(14444) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 956696) 14(14444) TOTAL: 6 free fragments = 957832 free bytes 14(14444) ----------------------------- 15(14445) INFO: signal 15 received 15(14445) Memory status (pkg): 15(14445) fm_status (0x8130780): 15(14445) heap size= 1048576 15(14445) used= 72048, used+overhead=90744, free=957832 15(14445) max used (+overhead)= 90744 15(14445) dumping free list: 15(14445) hash = 2 fragments no.: 1, unused: 0 bucket size: 16 - 16 (first 16) 15(14445) hash = 16 fragments no.: 3, unused: 0 bucket size: 128 - 128 (first 128) 15(14445) hash = 92 fragments no.: 1, unused: 0 bucket size: 736 - 736 (first 736) 15(14445) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 956696) 15(14445) TOTAL: 6 free fragments = 957832 free bytes 15(14445) ----------------------------- 16(14446) INFO: signal 15 received 16(14446) Memory status (pkg): 16(14446) fm_status (0x8130780): 16(14446) heap size= 1048576 16(14446) used= 72048, used+overhead=90744, free=957832 16(14446) max used (+overhead)= 90744 16(14446) dumping free list: 16(14446) hash = 2 fragments no.: 1, unused: 0 bucket size: 16 - 16 (first 16) 16(14446) hash = 16 fragments no.: 3, unused: 0 bucket size: 128 - 128 (first 128) 16(14446) hash = 92 fragments no.: 1, unused: 0 bucket size: 736 - 736 (first 736) 16(14446) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 956696) 16(14446) TOTAL: 6 free fragments = 957832 free bytes 16(14446) ----------------------------- 17(14447) INFO: signal 15 received 17(14447) Memory status (pkg): 17(14447) fm_status (0x8130780): 17(14447) heap size= 1048576 17(14447) used= 72048, used+overhead=90744, free=957832 17(14447) max used (+overhead)= 90744 17(14447) dumping free list: 17(14447) hash = 2 fragments no.: 1, unused: 0 bucket size: 16 - 16 (first 16) 17(14447) hash = 16 fragments no.: 3, unused: 0 bucket size: 128 - 128 (first 128) 17(14447) hash = 92 fragments no.: 1, unused: 0 bucket size: 736 - 736 (first 736) 17(14447) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 956696) 17(14447) TOTAL: 6 free fragments = 957832 free bytes 17(14447) ----------------------------- 18(14448) INFO: signal 15 received 18(14448) Memory status (pkg): 18(14448) fm_status (0x8130780): 18(14448) heap size= 1048576 18(14448) used= 72048, used+overhead=90744, free=957832 18(14448) max used (+overhead)= 90744 18(14448) dumping free list: 18(14448) hash = 2 fragments no.: 1, unused: 0 bucket size: 16 - 16 (first 16) 18(14448) hash = 16 fragments no.: 3, unused: 0 bucket size: 128 - 128 (first 128) 18(14448) hash = 92 fragments no.: 1, unused: 0 bucket size: 736 - 736 (first 736) 18(14448) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 956696) 18(14448) TOTAL: 6 free fragments = 957832 free bytes 18(14448) ----------------------------- 19(14449) INFO: signal 15 received 19(14449) Memory status (pkg): 19(14449) fm_status (0x8130780): 19(14449) heap size= 1048576 19(14449) used= 72048, used+overhead=90744, free=957832 19(14449) max used (+overhead)= 90744 19(14449) dumping free list: 19(14449) hash = 2 fragments no.: 1, unused: 0 bucket size: 16 - 16 (first 16) 19(14449) hash = 16 fragments no.: 3, unused: 0 bucket size: 128 - 128 (first 128) 19(14449) hash = 92 fragments no.: 1, unused: 0 bucket size: 736 - 736 (first 736) 19(14449) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 956696) 19(14449) TOTAL: 6 free fragments = 957832 free bytes 19(14449) ----------------------------- 1(14431) TOTAL: 4 free fragments = 1007824 free bytes 1(14431) ----------------------------- 0(14430) XLOG: destroy module ... 0(14430) pool_remove: Connection still kept in the pool 0(14430) pool_remove: Connection still kept in the pool 0(14430) pool_remove: Removing connection from the pool 0(14430) DEBUG: tm_shutdown : start 0(14430) DEBUG: unlink_timer_lists : emptying DELETE list 0(14430) DEBUG: tm_shutdown : emptying hash table 0(14430) DEBUG: tm_shutdown : releasing timers 0(14430) DEBUG: tm_shutdown : removing semaphores 0(14430) DEBUG: tm_shutdown : destroying tmcb lists 0(14430) DEBUG: tm_shutdown : done 0(14430) tls_close: Closing SSL connection 0(14430) tls_update_fd: New fd is 36 0(14430) tls_shutdown: Shutdown successful 0(14430) tls_tcpconn_clean: Entered 1(14431) dumping free list:

2 1

[Users] WeSIP
by Helmut Kuper 02 Mar '07

02 Mar '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I'm interessting in using WeSIP as a ALG. I got running openser as a SIP frontend for WeSIP by using SEAS module. But I have no idea what do to to get WeSIP running. Do I need here apache to run those xml files? Or is there any WeSIP binaries to strat WeSIP Server ?? regards Helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD4DBQFF6Bc84tZeNddg3dwRAmchAKCtliDoToB1SeWeNEJtQ/AKf4viJgCXVZPv D6k7Vzrru5NaXEXLL70IBQ== =kpT1 -----END PGP SIGNATURE-----

2 1

[Users] WeSIP API
by raviprakash sunkara 02 Mar '07

02 Mar '07

Hello Users Good Morning, Is Any Using the WeSIP to OpenSER for SEAS, Please can anyone Give the WeSIP API documentation , So that I view the Classes and Methods in WeSIP, Does WeSIP supports the b2bua for the OpenSER, So that I can Generate it, I'll use the WeSIP only for INVITE request, For Post Paid Billing Generating Main Question is WeSIP support the NAT, -- Thanks and Regards Ravi Prakash Sunkara ravi.sunkara(a)hyperion-tech.com M:+91 9985077535 www.hyperion-tech.com Client and Parent company :- www.august-networks.com

3 2

[Users] Some precisions about reply_route handling
by Jean-Franois SMIGIELSKI 01 Mar '07

01 Mar '07

Hello list, My openser (1.1.0) does not forward provisionnal replies. Currently, the "180 Ringing" I receive are absorbed, and that is bad. But I don't use any configuration directive to explicitely and statefully forward provisionnal answers. Please, which can I use? The reply_route block that is used is set with a call to t_on_reply . Thanks a lot. JF Smigielski.

2 1

[Users] Loose routing and branches
by Marius Patea 01 Mar '07

01 Mar '07

Hello I have an OpenSER which receives an initial INVITE request with a Route HF. It processes Route HF (by applying loose routing mechanism). Then it appends a new branch and relay the request. My problem is that the CANCEL request, which is generated by the TM module when a 200 OK response from the other leg was acknowledged, contains the original Route HF from the INVITE request, and not the processed Route HF. I think that the correct behavior is that the Route HF from the CANCEL request should not contains this proxy as topmost route. Thanks for any hints! Regards, MArius

1 0

[Serusers] SER as Presence Server
by gauri 01 Mar '07

01 Mar '07

Hi, We need Presence Server for testing purpose. Which version of SER has support for Presence Server and is it stable. Any pointers in this direction will be helpful. Thanks, Gauri 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

2 1

[Serusers] how to log unauthorized attemp with clear password
by Fabio Macchi 01 Mar '07

01 Mar '07

Hi all, I'm using ser-0.9.7 with mysql authentication. This is the INVITE handling section of my ser.cfg # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- if (nat_uac_test("19")) { setflag(6); } if (!allow_trusted()) { if (!proxy_authorize("sip.keeptelecom.com","subscriber")) { proxy_challenge("sip.keeptelecom.com","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); }; # ------ Logging calls after succesfully auth ----------- if (isflagset(6)) { xlog("L_INFO", "<%is> CH <%ct> R-URI <%ru> autenticato: NAT detected\n"); } else { xlog("L_INFO", "<%is> CH <%ct> R-URI <%ru> autenticato: no NAT detected\n"); } As you cas see, I've added just after authentication section some lines to log every succesfully authenticated attemp to make a call, with NAT detection info. I'd like, in the same way, add a single line to log every wrong user or wrong password attempt. My question is: which is the correct place to set this in the configuration file ? My first idea was to place xlog instruction just inside the if(!proxy_authorize){ . , but this produces multiple line in log, because in a call first invite is sent without proxy_authorization section. Is there a way, with xlog, to show the client sent password in clear for an easy troubleshoot ? Many thanks for any suggestion. Fabio

2 1

[Users] connection pooling with unixodbc
by Papadopoulos Georgios 01 Mar '07

01 Mar '07

hello all I am loading the following 10 modules that use an oracle database through unixodbc. modparam("usrloc|acc", "db_url", "unixodbc://user:pass@/OracleODBC-10g") modparam("auth_db|alias_db|uri_db|group|permissions|lcr|domain", "db_url", "unixodbc://user:pass@/OracleODBC-10g") modparam("avpops", "avp_url", "unixodbc://user:pass@/OracleODBC-10g") In my config I have fork=yes and tcp is enabled With 5 listen addresses and children=1 : 10 connections in the database. children=2 : 16 connections children=3 : 22 connections With 4 listen address and children=1 : 9 connections in the database. children=2 : 14 connections children=3 : 19 connections Could anyone help me understand the relation between listen addresses, number of children and number of modules that use the db? Also these numbers are the same whether I use MySQL both through the mysql module and the unixodbc module. Looking into the code, it seems like there is connection pooling into the unixodbc module. However I get all these connections. What am I doing wrong? thank you for any help George Disclaimer The information in this e-mail and any attachments is confidential. It is intended solely for the attention and use of the named addressee(s). If you are not the intended recipient, or person responsible for delivering this information to the intended recipient, please notify the sender immediately. Unless you are the intended recipient or his/her representative you are not authorized to, and must not, read, copy, distribute, use or retain this message or any part of it. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.

2 1

[Serusers] voice transmitting
by zahra bahar 01 Mar '07

01 Mar '07

Hi all. I have 2 xlite phone with ser server. when establshing call only for example user3 can hear user4 voice and user4 can hear the voice of itself too!and user3 should be on speakerphone mode. what's wrong?when we should be on speaker phone? --------------------------------- The fish are biting. Get more visitors on your site using Yahoo! Search Marketing.

1 0

Re: [Users] TLS and dead tcp child
by Paulo Angonese 01 Mar '07

01 Mar '07

The core, binaries and the full log are in ftp://ftp.procergs.com.br/pub/procergs/openser This is the backtrace: .... #0 0x00c63b87 in strchr () from /lib/libc.so.6 #1 0x0033dc53 in krb5_kt_resolve () from /usr/lib/libkrb5.so.3 #2 0x003cf5a0 in kssl_keytab_is_available () from /lib/libssl.so.5 #3 0x003bbc4c in ssl3_choose_cipher () from /lib/libssl.so.5 #4 0x003b73c0 in ssl3_accept () from /lib/libssl.so.5 #5 0x003c4eba in SSL_accept () from /lib/libssl.so.5 #6 0x080e1dc9 in tls_accept (c=0xb616b490) at tls/tls_server.c:236 #7 0x080e35cc in tls_fix_read_conn (c=0xb616b490) at tls/tls_server.c:872 #8 0x0809c428 in tcp_read_req (con=0xb616b490, bytes_read=0xbfad8448) at tcp_read.c:411 #9 0x0809c7a7 in handle_io (fm=0x814e1b0, idx=-1) at tcp_read.c:772 #10 0x0809dbf3 in io_wait_loop_epoll (h=0x8113020, t=Variable "t" is not available. ) at io_wait.h:722 #11 0x0809df67 in tcp_receive_loop (unix_sock=17) at tcp_read.c:893 #12 0x08093969 in tcp_init_children (chd_rank=0x810dcd4) at tcp_main.c:1745 #13 0x0806ce1e in main_loop () at main.c:1009 #14 0x0806e8cb in main (argc=1, argv=0xbfad86f4) at main.c:1477 My version of krb5-libs is krb5-libs-1.4.1-5 Please, any help? Thanks Paulo Angonese >>> >>> Paulo Angonese wrote: >>>> Ok. >>>> >>>> The core (two files was generated) and the full log (debug=9) are in >>>> ftp://ftp.procergs.com.br/pub/procergs/openser >>>> >>>> >>>> >>>> Bogdan-Andrei Iancu escreveu: >>>>> Hi Paulo, >>>>> >>>>> some suggestion :) - could you provide full log (debug) and >>>>> core+binaries (for download) ? Or backtrace? >>>>> >>>>> regards, >>>>> bogdan >>>>> >>>>> Paulo Angonese wrote: >>>>>> >>>>>> Hi, >>>>>> >>>>>> I have the same problem. >>>>>> When the SNOM ( 360 SIP 6.5.2 ) sends the REGISTER using TLS >>>>>> OpenSER crashes. >>>>>> I´m using >>>>>> OpenSer (1.1.1-tls (i386/linux)), >>>>>> openssl-0.9.7f-7.10 >>>>>> openssl-devel-0.9.7f-7.10 >>>>>> Fedora 2.6.11-1.1369_FC4 >>>>>> >>>>>> Any help? >>>>>> >>>>>> Thanks >>>>>> >>>>>> Paulo Angonese >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> [Users] CVS 31.03, TLS and dead tcp child >>>>>> Bogdan-Andrei Iancu bogdan at voice-system.ro >>>>>> Tue Oct 31 17:13:43 CET 2006 >>>>>> >>>>>> * Previous message: [Users] CVS 31.03, TLS and dead tcp child >>>>>> * Next message: [Users] Q:ABOUTE:SET UP MESSAGE FLOW >>>>>> * Messages sorted by: [ date ] [ thread ] [ subject ] [ author ] >>>>>> >>>>>> Hi Frank, >>>>>> >>>>>> can you pack the cores + binaries (lib + exe) + sources+logs(in >>>>>> debug >>>>>> mode) and put it somewhere for download? >>>>>> I will try to take a look. >>>>>> >>>>>> regards, >>>>>> Bogdan >>>>>> >>>>>> Frank DInnocenzo wrote: >>>>>> >>>>>> > Bogdan, >>>>>> > >>>>>> > I am using openser-1.1.0-tls_src. I actually get two core files >>>>>> after >>>>>> > the crash. Can I post them somehow? >>>>>> > >>>>>> > >>>>>> > # openser -V >>>>>> > version: openser 1.1.0-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. >>>>>> > @(#) $Id: main.c,v 1.20 2006/07/04 17:25:54 bogdan_iancu Exp $ >>>>>> > main.c compiled on 14:27:50 Oct 24 2006 with gcc 4.0.0 >>>>>> > >>>>>> > Thanks >>>>>> > Frank >>>>>> > >>>>>> > >>>>>> > */Bogdan-Andrei Iancu <bogdan at voice-system.ro>/* wrote: >>>>>> > >>>>>> > Hi Frank, >>>>>> > >>>>>> > I'm successfully using a SNOM 320 with TLS and no problem so >>>>>> far..... >>>>>> > What openser version are you using?...do you get any core file >>>>>> > after the >>>>>> > crash?? >>>>>> > >>>>>> > regards, >>>>>> > bogdan >>>>>> > >>>>>> > Frank DInnocenzo wrote: >>>>>> > >>>>>> > > Hello, >>>>>> > > >>>>>> > > I'm new to openSer and wondered if there was any >>>>>> resolution to this >>>>>> > > post??? I'm hitting this same crash with openSer (with TLS >>>>>> enabled) >>>>>> > > using a Snom phone (300 with latest Beta firmware - >>>>>> > > snom300-6.5-SIP-j.bin ). I've gone >>>>>> > > through the archives looking for help getting this setup >>>>>> > working. I've >>>>>> > > also tried the Snom Softphone with the same result. >>>>>> > > >>>>>> > > Thanks >>>>>> > > Frank >>>>>> > > >>>>>> > > >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>> >>>>>> > > >>>>>> > > Hi, >>>>>> > > >>>>>> > > I have tried to register to OpenSER CVS 31.03 with my SNOM >>>>>> 360 >>>>>> > > (5.3.6sw), disabled pretty all modules from OpenSER (radius, >>>>>> > xlog, auth, >>>>>> > > ...) and always the server crashes. >>>>>> > > >>>>>> > > Is there something I have missed in configuration ? >>>>>> > > >>>>>> > > Thanks a lot for help, >>>>>> > > -Mika >>>>>> > > >>>>>> > > The error message and debug info seems like this: >>>>>> > > >>>>>> > > 11(24351) tcpconn_new: new tcp connection to: XXX.XX.XX.XXX >>>>>> > > 11(24351) tcpconn_new: on port 2171, type 3 >>>>>> > > 11(24351) tls_tcpconn_init: Entered: Creating a whole new ssl >>>>>> > connection >>>>>> > > 11(24351) tls_tcpconn_init: Looking up tls domain >>>>>> > [XXX.XX.XX.XXX:5061] >>>>>> > > 11(24351) tls_tcpconn_init: Using default tls server settings >>>>>> > > 11(24351) tls_tcpconn_init: Setting in ACCEPT mode (server) >>>>>> > > 11(24351) tcpconn_add: hashes: 442, 1 >>>>>> > > 11(24351) handle_new_connect: new connection: 0xb608f6a0 24 >>>>>> > flags: 0002 >>>>>> > > 11(24351) send2child: to tcp child 0 7(24347), 0xb608f6a0 >>>>>> > > 7(24347) received n=4 con=0xb608f6a0, fd=19 >>>>>> > > 7(24347) DBG: io_watch_add(0x810e580, 19, 2, 0xb608f6a0), >>>>>> fd_no=1 >>>>>> > > 7(24347) tls_update_fd: New fd is 19 >>>>>> > > 11(24351) DBG: handle_tcp_child: dead tcp child 0 (pid >>>>>> 24347, no 7) >>>>>> > > (shutting down?) >>>>>> > > 11(24351) DBG: io_watch_del (0x810e420, 17, -1, 0x0) >>>>>> fd_no=16 called >>>>>> > > 11(24351) ERROR: receive_fd: EOF on 15 >>>>>> > > 11(24351) DBG: handle_ser_child: dead child 7, pid 24347 >>>>>> > (shutting down?) >>>>>> > > 11(24351) DBG: io_watch_del (0x810e420, 15, -1, 0x0) >>>>>> fd_no=15 called >>>>>> > > 0(24339) child process 24347 exited by a signal 11 >>>>>> > > 0(24339) core was generated >>>>>> > > 0(24339) INFO: terminating due to SIGCHLD >>>>>> > > 6(24346) INFO: signal 15 received >>>>>> > > 6(24346) Memory status (pkg): >>>>>> >>>>>> _______________________________________________ >>>>>> Users mailing list >>>>>> Users(a)openser.org >>>>>> http://openser.org/cgi-bin/mailman/listinfo/users >>>>>> >>>>> >>>>> >>>>> >>>> >>>> >>> >>> >>> >> >> >

3 2

[Users] FreeBSD port update to 1.1.1 committed
by Jesus Rodriguez 01 Mar '07

01 Mar '07

Hello, For all that people asking me about the FreeBSD port version :) With a very long delay (sorry), today has been committed to the FreeBSD ports tree the update to OpenSER 1.1.1 version. I hope to have the update to 1.2 version ready and committed as soon as it is realesed. Regards, Saludos JesusR. ------------------------------------ Jesus Rodriguez VozTelecom Sistemas, S.L. jesusr(a)voztele.com http://www.voztele.com Tel. 902360305 -------------------------------------

1 0

[Serusers] RTPProxy codecs?
by Andrey Kuprianov 01 Mar '07

01 Mar '07

Hi, It just struck me. Does RTPProxy operate only with certain codecs or does it proxy RTP regardless the codec? I was just compiling RTPProxy code from CVS and saw this *********************************************************************************** * Please contact sales(a)sippysoft.com if you need G.729 support in makeann * *********************************************************************************** Regards, Andrey.

2 2

[Serusers] enum module error ser2.0
by tzieleniewski 01 Mar '07

01 Mar '07

Hi! I was trying to utilize the enum module of ser but whenever I try to do the check for the user part of the request uri I get the following error: ser@sen:~/ser/ser-cvs/proxy$ ./ser -c -f ser.cfg 0(5401) parse error (721,14-15): unknown command, missing loadmodule? I have the enum module loaded and the line for this happens is: if (is_user_from_e164()) { I also tried is_e164() which I found in the module header file but with the same result. it this is not going to work how can I make the check by hand? for instance having such uri sip:+4822102@domain.com will somthing like this work: uri=~ "sip:+[0-9]{7}@.*" ?? Bests Tomasz

2 2

Re[4]: [Users] OpenSER+ Freeradius+MySQL
by Ion Minzu 01 Mar '07

01 Mar '07

Hello raviprakash, 1. FreeRADIUS Version 1.1.3 2. I have: 1st PC 2nd PC OpenSER FreeRadius radiusclient-ng MySQL Am have radiusclient-ng. 3. I want total AAA. > Hello Ion, > Let me know, Which version using for freeradius > U want integrating the OpenSER + radius with MySQl > OpenSER nerver communicate with Radius, For that U need the RadiusClient-ng for that..... OpenSER--------->>radiusclient------>radius server > Let me know . What U want real with that...total AAA > On 3/1/07, Ion Minzu <ion.minzu(a)cts.md> wrote: > Salut, > I'm a beginner in radius, in the DB radius I have tables: nas, > radacct, radcheck, radgroupcheck, radgroupreply, radpostauth, > radreply, usergroup. I need to add users in DB (in the file > /etc/raddb/users is troublesome for me because I have 300 users > of the service). radacct is similar to table acc from openser, > but I don't understand what table from DB radius is similar to > table subscriber from DB openser. >> Salut, >>> Hello , >>> I have 2 servers: 1st is OpenSER, second is FreeRadius with MySQL. >>> Till now the authentication was making through mysql, but from now I >>> want make authentication and accounting through radius. In mysql are 2 >>> DB: radius and openser. >>> -To create a new user, in which DB it will be inserted? >>> >> you should add it in the DB used by Radius server >>> -Till now I add new users with openserctl, but with radius how to add >>> new users? >>> >> I'm not a radius user, but I guess you can use any DB tool, including >> mysql client ;). >> Regards, >> Bogdan > -- > Best regards, > Ion Minzu, > mailto:ion.minzu@cts.md > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users -- Best regards, Ion Minzu, mailto:ion.minzu@cts.md

1 0

[Users] OpenSER+ Freeradius+MySQL
by Ion Minzu 01 Mar '07

01 Mar '07

Hello , I have 2 servers: 1st is OpenSER, second is FreeRadius with MySQL. Till now the authentication was making through mysql, but from now I want make authentication and accounting through radius. In mysql are 2 DB: radius and openser. -To create a new user, in which DB it will be inserted? -Till now I add new users with openserctl, but with radius how to add new users? -- Best regards, Ion Minzu, mailto:ion.minzu@cts.md

3 4

[Users] Multiple hops Accounting
by Manuel Guesdon 28 Feb '07

28 Feb '07

Hi, I try to have exploitable accounting for multiple domains and multi hops routing. I've previously tried for weeks on asterisk without success, even with adding c modules and playing with cdr fork :-( He is a case: - sip.domainA and sip.domainB point to the same Openser server. - 4(a)sip.domainA is dbaliased to 5(a)sip.domainB - 5(a)sip.domainB is dbaliased to 6@asteriskServer - 6@asteriskServer make a dial to phone2(a)sip.domainA I dial 4 on phone1 and get this INVITE routing phone1(a)sip.domainA === RURI:4@sip.domainA ==> openser openser === RURI:6@asteriskServer ==> asteriskServer asteriskServer == RURI:phone2@sip.domainA ==> openser openser ==> phone2 this works and phone2 ring, I take the call and hangup phone1 after some time. Now in acc table I find 15 records (7 INVITEs, 5 ACKs and 3 BYEs) The 3 BYE records seems correct (one from phone 1 to openser, one from openser to asteriskBox and the last from asteriskBox to phone 2 via openser). But I can't find a way to match the 7 INVITEs to the 3 BYEs, even with adding src&dst legs src_ip info and so on. >From acc records, I'd like to have make the following records report: == Call ID, src, dst, total duration, billing duration == - 1234, phone1@sip.domainA,4@sip.domainA,10,8 - 1234, 4@sip.domainA,5@sip.domainB,10,8 - 1234, 5@sip.domainB,6@asteriskBox,10,8 - 1234, 6@asteriskBox,phone2@domainA,10,8 Is it a complete dream or do you have an idea ? Manuel

2 1

[Serusers] problem in calling voicemail!!
by Meidiana Ten 28 Feb '07

28 Feb '07

Hi.. Thanx for the solution It really works well But now I face another problem Really need your help.. I configure SER for voicemail usage I'm using eyebeam as the UA. To check voicemail, I set it to call 113, and to send voicemail, it will call 114 based on the SER script. But it failed to forward to voicemail even the icon "send to voicemail" has appeared on the eyebeam. Below is my ser.cfg and ser_sems.cfg Please check it for me and tell me what's wrong. ser.cfg: debug=9 fork=yes log_stderror=yes listen=202.95.149.2 # put your server IP address here port=5060 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:heslo@localhost/ser" fifo_mode=0666 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/nathelper.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/avpops.so" loadmodule "/usr/local/lib/ser/modules/domain.so" loadmodule "/usr/local/lib/ser/modules/permissions.so" loadmodule "/usr/local/lib/ser/modules/msilo.so" modparam("auth_db|permissions|uri_db|usrloc","db_url", "mysql://ser:heslo@localhost/ser") modparam("auth_db|uri_db|usrloc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_sock", "/var/run/rtpproxy.sock") modparam("usrloc", "db_mode", 2) modparam("registrar", "nat_flag", 6) modparam("rr", "enable_full_lr", 1) modparam("tm", "fr_inv_timer", 27) modparam("tm", "fr_inv_timer_avp", "inv_timeout") modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") modparam("tm", "fr_timer", 10 ) modparam("tm", "fr_inv_timer", 15 ) modparam("tm", "wt_timer", 10 ) modparam("msilo", "db_url", "mysql://ser:heslo@localhost/ser") modparam("msilo", "db_table", "silo") modparam("msilo","registrar","sip:registrar@pcr.ac.id") modparam("msilo","expire_time",259200) modparam("msilo","check_time",30) modparam("msilo","clean_period",5) alias="pcr.ac.id" route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); break; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); break; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method!="REGISTER") { record_route(); }; if (method=="BYE" || method=="CANCEL") { unforce_rtp_proxy(); } # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if (has_totag() && (method=="INVITE" || method=="ACK")) { if (nat_uac_test("19")) { setflag(6); force_rport(); fix_nated_contact(); }; force_rtp_proxy("l"); }; route(1); break; }; # ----------------------------------------------------------------- # Offline Message Store Section # ----------------------------------------------------------------- if (is_from_local()) { if (method=="REGISTER") { save("location"); log("REGISTER received -> dumping messages with MSILO\n"); # MSILO - dumping user's offline messages if (m_dump()) { log("MSILO: offline messages dumped - if they were\n"); }else{ log("MSILO: no offline messages dumped\n"); }; break; }; # domestic SIP destinations are handled using our USRLOC DB if(!lookup("location")) { if (! t_newtran()) { sl_reply_error(); break; }; # we do not care about anything else but MESSAGEs if (!method=="MESSAGE") { if (!t_reply("404", "Not found")) { sl_reply_error(); }; break; }; log("MESSAGE received -> storing using MSILO\n"); # MSILO - storing as offline message if (m_store("0")) { log("MSILO: offline message stored\n"); if (!t_reply("202", "Accepted")) { sl_reply_error(); }; }else{ log("MSILO: offline message NOT stored\n"); if (!t_reply("503", "Service Unavailable")) { sl_reply_error(); }; }; break; }; # if the downstream UA does not support MESSAGE requests # go to failure_route[1] t_on_failure("1"); t_relay(); break; }; # forward to current uri now; use stateful forwarding that # works reliably even if we forward from TCP to UDP if (!t_relay()) { sl_reply_error(); }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(5); route(1); break; }; if (uri==myself) { if (method=="ACK") { route(6); break; } else if (method=="CANCEL") { route(3); break; } else if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { route(5); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; }; route(1); } failure_route[1] { # forwarding failed -- check if the request was a MESSAGE if (!method=="MESSAGE") { break; }; log(1,"MSILO:the downstream UA doesn't support MESSAGEs\n"); # we have changed the R-URI with the contact address, ignore it now if (m_store("1")) { log("MSILO: offline message stored\n"); t_reply("202", "Accepted"); }else{ log("MSILO: offline message NOT stored\n"); t_reply("503", "Service Unavailable"); }; } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" && isflagset(6)) { unforce_rtp_proxy(); }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- if (!search("^Contact:\ +\*") && nat_uac_test("19")) { setflag(6); fix_nated_register(); force_rport(); }; sl_send_reply("100", "Trying"); if (!www_authorize("pcr.ac.id","subscriber")) { www_challenge("pcr.ac.id","0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # CANCEL and INVITE Message Handler # ----------------------------------------------------------------- if (!allow_trusted() && nat_uac_test("19")) { setflag(6); } lookup("aliases"); if (method=="INVITE" && !allow_trusted()) { if (!proxy_authorize("pcr.ac.id","subscriber")) { proxy_challenge("pcr.ac.id","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); }; if (uri =~ "sip:1.*@") { log("Forwarding to 5070"); rewritehostport("localhost:5070"); t_relay_to_udp("localhost","5070"); break; } if (uri=~"^sip:9[0-9]*@") { route(4); break; }; if (uri!=myself) { route(5); route(1); break; }; if (!lookup("location")) { if (uri=~"^sip:[0-9]{10}@") { route(4); break; }; sl_send_reply("404", "User Not Found"); break; }; if (isflagset(6)) { force_rport(); fix_nated_contact(); force_rtp_proxy(); }; t_on_reply("1"); if (!t_relay()) { if(isflagset(6)) { unforce_rtp_proxy(); } sl_reply_error(); }; } route[4] { # ----------------------------------------------------------------- # PSTN Handler # ----------------------------------------------------------------- rewritehostport("202.95.149.2:5060"); # INSERT YOUR PSTN GATEWAY IP ADDRESS avp_write("i:45", "inv_timeout"); if (isflagset(6)) { force_rport(); fix_nated_contact(); force_rtp_proxy(); }; route(1); } onreply_route[1] { if (isflagset(6) && status=~"(180)|(183)|2[0-9][0-9]") { if (!search("^Content-Length:\ +0")) { force_rtp_proxy(); }; }; if (nat_uac_test("1")) { fix_nated_contact(); }; } ser_sems.cfg: # ser.cfg,v 1.25.2.1 2005/02/18 14:30:44 andrei Exp $ # # ser for sems configuration # ----------- global configuration parameters ------------------------ debug=9 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=yes # (cmd line: -E) listen=202.95.149.2 port=5070 children=4 #check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:heslo@localhost/ser" fifo_mode=0666 unix_sock="/tmp/ser_sock" # ------------------ module loading ---------------------------------- 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/textops.so" loadmodule "/usr/local/lib/ser/modules/avp.so" loadmodule "/usr/local/lib/ser/modules/avpops.so" # ----------------- setting module-specific parameters --------------- alias="pcr.ac.id" # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # configure avpops db connection #modparam( "avpops", "avp_url", "mysql://ser:heslo@localhost/ser" ) #modparam( "avpops", "avp_table", "subscriber" ) #modparam( "avpops", "uuid_column", "id" ) # configure aliases, the number doesn't matter as long as there are no collisions) modparam( "avpops", "avp_aliases", "email=i:67" ) # scheme to access the database #modparam( "avpops", "db_scheme", # "email_scheme:table=subscriber;value_col=email_address;value_type=string") #modparam("tm", "pass_provisional_replies", 1) # appends for INVITE to voicemail modparam("tm", "tw_append","voicemail_headers:hdr[User-Agent];P-Email-Address=avp[$email]") # appends for dtmf per INFO modparam( "tm", "tw_append","info_append:hdr[Content-Length];hdr[subject];msg[body]") # 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 > max_len ) { sl_send_reply("513", "Message too big"); break; }; if (method != "ACK" && method != "INVITE" && method != "BYE" && method != "CANCEL" && method != "INFO" ){ log("unsupported method\n"); sl_send_reply("500","unsupported method"); 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 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"); if (!t_relay()) { sl_reply_error(); } break; }; # make transaction if (!t_newtran()){ log("could not create transaction\n"); sl_send_reply("500","could not create transaction"); break; } # actively absorb ACKs if (method == "ACK") { t_relay(); break; } # pass INFO to SEMS if (method=="INFO") { if(!t_write_unix("/tmp/sems_sock","sems/info_append")){ log("could not contact sems\n"); t_reply("500","could not contact media server"); } } if (uri =~ "sip:101.*@") { if (!t_write_unix("/tmp/sems_sock","myapp")){ log("could not contact media server\n"); t_reply("500","could not contact media server"); break; } break; } if (uri =~ "sip:102.*@") { if (!t_write_unix("/tmp/sems_sock","myconfigurableapp")){ log("could not contact media server\n"); t_reply("500","could not contact media server"); break; } break; } if (uri =~ "sip:103.*@") { if (!t_write_unix("/tmp/sems_sock","myannounceapp")){ log("could not contact media server\n"); t_reply("500","could not contact media server"); break; } break; } if (uri =~ "sip:104.*@") { if (!t_write_unix("/tmp/sems_sock","myjukebox")){ log("could not contact media server\n"); t_reply("500","could not contact media server"); break; } break; } if (uri =~ "sip:105.*@") { if (!t_write_unix("/tmp/sems_sock","mycc")){ log("could not contact media server\n"); t_reply("500","could not contact media server"); break; } break; } if (uri =~ "sip:106.*@") { if (!t_write_unix("/tmp/sems_sock","ivr_announce")){ log("could not contact media server\n"); t_reply("500","could not contact media server"); break; } break; } ############ default sems apps if (uri =~ "sip:110.*@") { if (!t_write_unix("/tmp/sems_sock","echo")){ log("could not contact echo\n"); t_reply("500","could not contact media server"); break; } break; } if (uri =~ "sip:111.*@") { if (!t_write_unix("/tmp/sems_sock","announcement")){ log("could not contact announcement\n"); t_reply("500","could not contact media server"); break; } break; } if (uri =~ "sip:112.*@") { if (!t_write_unix("/tmp/sems_sock","conference")){ log("could not contact conference\n"); t_reply("500","could not contact media server"); break; } break; } if (uri =~ "sip:113.*@") { if (!t_write_unix("/tmp/sems_sock","mailbox")){ log("could not contact mailbox\n"); t_reply("500","could not contact media server"); break; } break; } if (uri =~ "sip:114.*@") { if (!t_write_unix("/tmp/sems_sock","early_announce")){ log("could not contact early_announce\n"); t_reply("500","could not contact media server"); break; } break; } if (uri =~ "sip:115.*@") { # load email avp with some email address avp_write("root@localhost","$email"); # use voicemail_headers append to pass it to sems if (!t_write_unix("/tmp/sems_sock","voicemail/voicemail_headers")){ log("could not contact voicemail\n"); t_reply("500","could not contact media server"); break; } break; } t_reply("404","Not found"); } Please tell me if you need my voicemail.conf..I'll send it to you. Thanx for the help Regards, Meidiana ____________________________________________________________________________________ We won't tell. Get more on shows you hate to love (and love to hate): Yahoo! TV's Guilty Pleasures list. http://tv.yahoo.com/collections/265

2 1

[Users] DB migration from 1.1 to 1.2
by Bogdan-Andrei Iancu 28 Feb '07

28 Feb '07

Hi, today I uploaded on SVN trunk the new logic from migrating the existing DBs from 1.1 format to 1.2. The strategy is different than in the previous version. The old "reinstall" proved to be suboptimal because of the lack of flexibility and mainly it didn't provide a smooth transition (downtime was required in order to swap the DBs, unsafe - data could be lost if error occurred during reinstall - and data could be lost because of DB changes). A new approach was chosen this time - migration. A new DB is created in parallel with the old one and data is processed and transferred into the new DB via sql commands (and nor dumps), using strict mapping between old and new columns. This way the original data is kept and complex column renaming and data processing is possible (like lcr address for example). for the moment this migration is done only for mysql, but is possible to do it for postgres also but as I'm not to much into, I will need some help from some postgres guru ;). in the mean while testing is welcomed ;) regards, bogdan

1 0

[Serusers] Call from Asterisk to SER to a foreign subscriber
by karim basraoui 28 Feb '07

28 Feb '07

---------- Forwarded message ---------- From: karim basraoui <basraouik(a)gmail.com> Date: 28 févr. 2007 20:33 Subject: Re: [Serusers] Call from Asterisk to SER to a foreign subscriber To: ram <talk2ram(a)gmail.com> I have found the thing! in Asterisk dial plan, when i'm trying to call my UA1, i should not do this: exten =>xxxx,Dial(SIP/UA1) but : exten =>xxxx,Dial(SIP/192.168.4.65/UA1) ;where 192.168.4.65 is the new host address thank you for your help ! 2007/2/24, karim basraoui <basraouik(a)gmail.com>: > > UA2 tries to call UA1 not UA3, sorry for the mistake > > 2007/2/24, karim basraoui < basraouik(a)gmail.com>: > > > > i had : > > > > UA1 (192.168.4.66:5061) > > > > }------------->Asterisk----------->SER-----------> UA3 ( > > 192.168.4.65:5061) > > UA2 ( 192.168.4.66:5062) > > > > then UA1 moved to SER domain : > > > > > > UA3 ( 192.168.4.65:5061) > > UA2 ( 192.168.4.66:5062) > > ------------->Asterisk----------->SER-----------> > > > > UA1 (192.168.4.66:5061) > > > > UA1 registers to SER and SER forwards the registration to ASTERISK, > > Asterisk is aware of the new location. > > > > UA2 tries to call UA3. It sends the invite to Asterisk and Asterisk > > replies with a TRYING message. > > after a while the call fails with an error > > > > > > 2007/2/24, ram < talk2ram(a)gmail.com>: > > > > > > then register with SER, should do the job > > > > > > UA----ASterisk----SER-----Phone > > > > > > is this correct > > > > > > ram > > > > > > > > > On 2/24/07, karim basraoui <basraouik(a)gmail.com > wrote: > > > > > > > > Hi, > > > > > > > > the call i'm trying to establish is from ASTERISK to SER. > > > > the UA i'm calling was initially in ASTERISK domain and moved to SER > > > > domain , a roaming case. > > > > > > > > 2007/2/24, ram < talk2ram(a)gmail.com>: > > > > > > > > > > Hi > > > > > > > > > > your question is not clear > > > > > > > > > > how is the call path happening from ASterisk or from SER > > > > > or both are gateway proxies ? > > > > > > > > > > ram > > > > > > > > > > > > > > > On 2/23/07, karim basraoui < basraouik(a)gmail.com > wrote: > > > > > > > > > > > Hi, > > > > > > > > > > > > I've set up a small architecture in which an ASterisk proxy is > > > > > > connected to a SER proxy in two different domains. > > > > > > > > > > > > I have sip phones connected to both of them. when an asterisk > > > > > > subscriber "travels" to SER domain it tries to register to SER. I've > > > > > > configured SER to accept foreign > > > > > > subscribers and to REGISTER them to their initial domain by the > > > > > > command "t_replicate". Till now everything works good, the UA is registered > > > > > > and Asterisk is aware > > > > > > of its new location, it displays in peers list the UA and its > > > > > > new IP address and port. > > > > > > > > > > > > The problem is when i'm trying to call the foreign UA from an > > > > > > Asterisk subscriber, it doesn't work. i have a TRYING response but no call > > > > > > establishement. > > > > > > > > > > > > how can i overcome this problem please? > > > > > > > > > > > > -- > > > > > > BASRAOUI Karim > > > > > > www.basraouik.afrikart.net > > > > > > _______________________________________________ > > > > > > Serusers mailing list > > > > > > Serusers(a)lists.iptel.org > > > > > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > > > > > > > > > > > > > > > > > > > > > > > > > -- > > > > BASRAOUI Karim > > > > www.basraouik.afrikart.net > > > > > > > > > > > > > > > > -- > > BASRAOUI Karim > > www.basraouik.afrikart.net > > > > > > -- > BASRAOUI Karim > www.basraouik.afrikart.net > -- BASRAOUI Karim www.basraouik.afrikart.net -- BASRAOUI Karim www.basraouik.afrikart.net

1 0

[Users] New OpenSER Config Generator features
by Andreas Granig 28 Feb '07

28 Feb '07

Hi all, Three new features for sip:wizard, the OpenSER Config Generator, have been released this week: - user-aliases: You can now assign aliases to local users. This is especially useful for mapping numeric numbers to SIP accounts for PSTN inward dialing. - cli-pai: This feature allows the provisioning of arbitrary SIP URIs, which are inserted into a P-Asserted-Identity Header for outgoing calls. This for example allows the signalling of a numeric CLI to a PSTN gateway (so it’s the opposite of user-aliases). - ring-timeout: You can now dynamically set the ring-timeout (or, more specifically, the fr_inv_timer) per local callee. Visit http://www.sipwise.com/wizard to give it a try. Regards, Andreas

3 6

Re: [Users] 3xx requests and t_relay
by Bogdan-Andrei Iancu 28 Feb '07

28 Feb '07

Hi Shane, have you followed this ex: http://www.openser.org/docs/modules/1.2.x/uac_redirect.html#AEN251 can you be more specific about the errors you see? regards, bogdan Shane Burrell wrote: > I am calling the t_relay after getting the contacts. However internally I > see in openser it makes it back to the route and then dies in loose route. > I assumed there was something I need to check to allow the new call to be > processed. > > -----Original Message----- > From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] > Sent: Wednesday, February 28, 2007 12:26 PM > To: Shane Burrell > Cc: users(a)openser.org > Subject: Re: [Users] 3xx requests and t_relay > > Hi, > > let me see if I get it right...you get a 3xx reply and try to generate > INVITE to the new contacts? if so, take a look at the uac_redirect module. > > regards, > bogdan > > Shane Burrell wrote: > >> I have this working to the point it tries to relay the call but I see >> that internally openser is not making the new call. I assume t_relay >> calls a new route but how to I make sure that all relays get >> forwarded. Some of my logic else where seems to be killing the >> rerouted call and I need any 302 routed calls to be accepted and routed. >> >> >> >> TIA, >> >> >> >> Shane >> >> >> *************************************************************** >> >> >> >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> Users mailing list >> Users(a)openser.org >> http://openser.org/cgi-bin/mailman/listinfo/users >> >> > > >

1 0

[Users] 3xx requests and t_relay
by Shane Burrell 28 Feb '07

28 Feb '07

I have this working to the point it tries to relay the call but I see that internally openser is not making the new call. I assume t_relay calls a new route but how to I make sure that all relays get forwarded. Some of my logic else where seems to be killing the rerouted call and I need any 302 routed calls to be accepted and routed. TIA, Shane ***************************************************************

2 1

RE: [Users] cross nat pinging
by Marcus Hunger 28 Feb '07

28 Feb '07

Hi Fatih, you could spoof the natping's source-ip so it would apear to come from the same host. I created a patch for that. It uses raw-sockets and works for me on linux. The patch is made against trunk. To use it, just set the udpping_from parameter of module nathelper. modparam("nathelper", "udpping_from" , "myhost:myport") Feedback on this would be welcome. Best regards, Marcus Fatih Cerit wrote: > Hi all > > I am testing nat pinging. I have a natted UA and 2 opensers with > diffrent ips. I decided to ping same nat from 2 opensers. I am going to > have ability to increase ping interval from 15 seconds to 30 > seconds. But I don't have any idea that cross pinging works or not. Is > there any one at there whom tested before or any suggestions?? > > Thanks > Fatih Cerit -- Marcus Hunger - hunger(a)sipgate.de Telefon: +49 (0)211-63 55 55-61 Telefax: +49 (0)211-63 55 55-22 indigo networks GmbH - Gladbacher Str. 74 - 40219 Düsseldorf HRB Düsseldorf 39841 - Geschäftsführer: Thilo Salmon, Tim Mois Steuernummer: 133/5838/1589, Umsatzsteuer-ID: DE219349391 www.sipgate.de - www.sipgate.at - www.sipgate.co.uk

2 1

[Users] AVPPOPS preformace question (OpenSER 1.1.x)
by Mike O'Connor 28 Feb '07

28 Feb '07

Hi Guys What is going to be faster: if (avp_db_load("$ruri/username", "$avp(s:callfwdimmediate)")) { .... }; if (avp_db_load("$ruri/username", "$avp(s:callfwdbusy)")) { .... }; if (avp_db_load("$ruri/username", "$avp(s:callfwdnoanswer)")) { .... }; or (if it works): avp_db_load("$ruri/username", "*"); if( is_avp_set("$avp(s:callfwdimmediate)")) { .... }; if( is_avp_set("$avp(s:callfwdbusy)")) { .... }; if( is_avp_set("$avp(s:callfwdnoanswer)")) { .... }; If the second was valid I would have thought it would be quicker. Thanks Mike

2 1

[Serusers] register_watcher(): Domain 'location' not found
by SER LIST 28 Feb '07

28 Feb '07

Hello, I am getting the message register_watcher(): Domain 'location' not found logged in my /var/log/messages file. Any idea what this is and to what it could be pointing to. I am running ser-2.0 daily-snapshop from Feb24. Thanks in advance _________________________________________________________________ Windows Live Spaces: share your New Year pictures! http://spaces.live.com/?mkt=en-ca

1 0

[Users] SIP Firewall
by Sergio G 28 Feb '07

28 Feb '07

Hello, I'm Sergio... Can anybody tell me if OpenSer could be used in the following way: ANY SIP REQUEST != REGISTER (to port 5060) ----------------------------> OpenSER --------> SOME PROXY_SIP (To port 5060) and the REGISTERs Request, redirected to another PROXY SIP and to another port... REGISTER REQUEST (to port 5060) --------------------------------> OPENSER------> REGISTRAR PROXY (to port 5062) I need to use it as some kind of Firewall SIP, REGISTER redirected to some IP and to some port, and the other requests redirected to another IP and port diferent. If this could be done, How? Where should I start reading? Thanks any help, Sergio

2 1

[Users] Multiple rtpproxy
by Ibrahim Hamouda 28 Feb '07

28 Feb '07

Hi Guys Could anybody give me a code snippet how to setup multiple rtpproxy in openser.cfg Thanks Ibrahim Hamouda

2 1

[Users] cancelling INVITE during OK response
by Web Monkey 28 Feb '07

28 Feb '07

hi, is there a way to cancel or terminate an INVITE or call upon receipt of the OK reply. my configuration invokes an external program which either return success or failure for OK message (response to INVITE). when failure is returned, i want the script to terminate the call or cancel the INVITE (OK message will not be forwarded to the calling UA). is there an existing function or module to realize this? the scenario is drawn below with provisional messages removed. UAC1 ------------------ proxy1 ----------------- proxy2 --------------- UAC2 call: INVITE ---------------------------> INVITE ----------------------> INVITE -----------------> OK <---------------- *XXX - error occurred during processing OK message thanks, sho --------------------------------- It's here! Your new message! Get new email alerts with the free Yahoo! Toolbar.

2 1

[Users] openserctl problem
by Helmut Kuper 28 Feb '07

28 Feb '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, After fixing that Query error I tried to add 2 user to openser. The first was successfully added to database. the 2nd failed with: ERROR 1062 (23000) at line 1: Duplicate entry '' for key 3 ERROR: introducing the new user '103(a)sip0.en.ewetel.de' to the database failed I looked into the database table subscribers and saw that column phplib_id is of type UNIQUE. I looked into the detailed mysql.log file and find out that the executed INSERT statement doesn't set the column phplib_id, so the 2nd user would have the same phplib_id value ('' or NULL) as the first. Because of column typ UNIQUE this will lead to that error above. Maybe someone of the developers can fix this. regards Helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFF5Etw4tZeNddg3dwRAiX3AJwKD0DDVwKDxbeX1/9fVitIxQSaKgCgliv5 u5IIkongIw3a16e4uu+vm4A= =kV49 -----END PGP SIGNATURE-----

2 1

[Users] openserctl error
by Helmut Kuper 28 Feb '07

28 Feb '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, the openserctl of the latest SVN trunk seems to has an error in the user add command. Line 1544 looks like this: UERY="insert into $SUB_TABLE ($ instead of QUERY="insert into $SUB_TABLE ($ regards helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFF5EdQ4tZeNddg3dwRAr7xAKCjGVi++E6eX3EhDVOj3Bm85FHSZACfd3OA 6xeUy4rZ7urCebfmALAjApQ= =93G4 -----END PGP SIGNATURE-----

2 1

[Users] OpenSER wiki semi-dead?
by Tais M. Hansen 28 Feb '07

28 Feb '07

Hi, No matter what subpage I try to access on the wiki, only the front page is displayed. Is anyone else seeing this? http://openser.org/dokuwiki/doku.php -- Regards, Tais M. Hansen ___________________________________________________________ http://www.linkedin.com/in/thansen

2 2

[Users] Auth with Radius+Kerberos
by Stefano Capitanio 28 Feb '07

28 Feb '07

Hi List, we are trying to do authentication with SER -> RADIUS -> KERBEROS following draft-kaushik-radius-sec-ext (Radius Secutity Extensions Using Kerberos) has anyone succeeded in doing this (or something similar)? thanks, Stefano

2 2

[Users] OpenSER 1.2.0
by Danilo Vilela Dantas 28 Feb '07

28 Feb '07

Hi all, We are working VoIP Project. We are using OpenSER 1.0.1. Unfortunately it doesn't support xcap and presence modules. Only the version 1.2.0 of the OpenSER. Is it ready openser 1.2.0? Or still delay? Thanks, regards, Danilo

4 3

[Users] cross nat pinging
by Fatih Cerit 28 Feb '07

28 Feb '07

Hi all I am testing nat pinging. I have a natted UA and 2 opensers with diffrent ips. I decided to ping same nat from 2 opensers. I am going to have ability to increase ping interval from 15 seconds to 30 seconds. But I don't have any idea that cross pinging works or not. Is there any one at there whom tested before or any suggestions?? Thanks Fatih Cerit

2 1

[Serusers] ser usrloc expiration
by Corey 28 Feb '07

28 Feb '07

The command "serctl ul show" displays an 'Expires' entry. I'm wondering: * what is the default expiration value? * where/how does this get set, and is it configurable? Many thanks!

2 1

[Users] TLS between UA and Openser
by Jayesh Nambiar 28 Feb '07

28 Feb '07

Hi everybody, I am trying to achieve TLS setup between the UAs and OpenSER. I am using PAP2 and SPA series for testing. Now the PAP2 does not support TLS transport, is it possible to use TLS setup between OpenSER and PAP2 using DNS-SRV? I tried using DNS-SRV but was not successful. The SPA supports TLS transport. I have generated the priviate key and required certs in the OpenSER. For SPA to work, how do we incorporate the certificate inside the SPA? Or should I get a certificate generated from Linksys and then add it to my CA list in OpenSER? I am not very familiar with the TLS concept but still learning. Thanks in advance. w/regards, Jayesh __________________________________________________________ Yahoo! India Answers: Share what you know. Learn something new http://in.answers.yahoo.com/

1 0

[Users] How do to prefix or pattern based routing
by Chris Heiser 28 Feb '07

28 Feb '07

It's more than possible there's a module or piece of documentation that I've been unable to uncover in my attempts to solve this issue. Basically, I'm looking for a way to do prefix or pattern based routing using a datasource (such as dbtext or a real database). I don't want to have to hack an openser.cnf file for each server I setup. That would just be silly. I've looked at PDT, but 1) it does source domain matching, which I could really care less about, and 2) it's misbehaving or misdocumented, as it won't allow for more than one prefix to be defined for a single source domain. I'd really love to be able to do pattern matches and route that way. But at this point, anything is better than hacking the config file. Any suggestions? --Chris

2 1

[Users] new MediaProxy release 1.8.2
by Adrian Georgescu 27 Feb '07

27 Feb '07

Hello, There is a new MediaProxy release available. To upgrade go to http://mediaproxy.ag-projects.com or download the software directly from: http://mediaproxy.ag-projects.com/mediaproxy-1.8.2.tar.gz Changes from version 1.8.1 to 1.8.2 ----------------------------------- - Added support to setup sessions by either caller or called as the initial INVITE may not contain a SDP body (Jeff Williams <jeffw(a)globaldial.com>) - Added support to update the RTP stream sign-in addresses after re- INVITES (Jeff Williams <jeffw(a)globaldial.com>) Changes from version 1.8.0 to 1.8.1 ----------------------------------- - Added User-Name attribute to the radius accounting Update request (freeradius needs it to be able to proxy the request) This release includes new functionality. Your feedback is appreciated, if you wish to provide it please reply to the mailing lists. Kind regards, Adrian Georgescu

3 8

[Users] web gui for registered openser user
by alper ozbilen 27 Feb '07

27 Feb '07

Dear All, I am new in openser. I have just installed openser on Fedora core 4 and run it. *I need "php user gui" for user to make web-based registeration process. I will able to bult it up but i dont know which mysql table i should you. *My aim is to give service to registered user on my datebase. In order to implement it, which open module should i install ? any help wolud be appreciated Thank a lot with my best regard. Alper Ozbilen _________________________________________________________________ Want a degree but can't afford to quit? Top school degrees online - in as fast as 1 year http://forms.nextag.com/goto.jsp?url=/serv/main/buyer/education.jsp?doSearc…

3 4

[Serusers] ser2 and sems?
by TZieleniewski 27 Feb '07

27 Feb '07

Hi!! I am going to set up the sems-ser configuration. In the mini-sems-ser-howto there is a clue to have separate instance of ser for sems. Will sems 0.10.0 work with the ser ottendorf release?? Bests tomasz

1 0

[Users] install:1.1.x-to-1.2.x
by Daniel-Constantin Mierla 27 Feb '07

27 Feb '07

(started a new thread from a devel discuttion) Everybody is encouraged to update the page below: http://openser.org/dokuwiki/doku.php/install:1.1.x-to-1.2.x If you do not know how to explain clear, but you know (feel) something changed from 1.1.x to 1.2.0 in a module or other place, just introduce in bold red a short phrase at the beginning or in module's section. Someone will see later and elaborate it. Cheers, Daniel On 02/27/07 13:01, Andreas Granig wrote: > Thank god there is > http://openser.org/dokuwiki/doku.php/install:1.1.x-to-1.2.x > > Hopefully, all the small changes are documented there (hint: the one > below isn't yet) ;o) > > Cheers, > Andreas > > Bogdan-Andrei Iancu wrote: >> Hi Juha, >> >> yes, there is an impact on the script, but very small. The only >> change is: >> modparam("registrar", "sip_natping_flag", x ) >> goes into >> modparam("nathelper","sipping_bflag",x) >> >> regards, >> bogdan

1 0

[Users] Presence module
by Helmut Kuper 27 Feb '07

27 Feb '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, I just configured the presence modul and got some strange output in my shell when my xlite phone trys to subscribe to a non existant user: 0(16615) ERROR: warning_builder: buffer size exceeded 0(16615) WARNING: warning skipped -- too big 0(16615) ERROR: warning_builder: buffer size exceeded 0(16615) WARNING: warning skipped -- too big 0(16615) ERROR: warning_builder: buffer size exceeded 0(16615) WARNING: warning skipped -- too big 0(16615) PRESENCE: get_subs_dialog:The query for subscribtion for [user]= sebastian.bruening%40googlemail.comtalk.v104112f54d0,[domain]= sip0.en.ewetel.de for [event]= presence.winfo returned no result 0(16615) PRESENCE:query_db_notify: Could not get subs_dialog from database 0(16615) PRESENCE:update_subscribtion:Could not send notify for presence.winfo 0(16615) PRESENCE:get_xcap_tree:The query in table xcap for [username]=sebastian.bruening%40googlemail.comtalk.v104112f54d0 , domain=sip0.en.ewetel.de returned no result any ideas what this mean? I use latest version of openser (1.2.0). regards Helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFF5D+D4tZeNddg3dwRAiYcAKCt9FJYu/yM6+HW/+QOXlp0XOAPWACfYpFZ 0JbJo+z27a2M1zr8WlEG4G4= =I9/o -----END PGP SIGNATURE-----

2 1

[Users] bug in radiusclient ng for 64b
by Stuart Clark 27 Feb '07

27 Feb '07

Hello, I have a 64bit machine running openser 1.1.0, and wish to log radius accounting details to radius using the acc module and radiusclient-ng. I am facing issues talking to a freeradius server, as described at: https://developer.berlios.de/bugs/?func=detailbug&bug_id=7442&group_id=1208 Basically, the server reports: Error: Received Accounting-Request packet from w.x.y.z with invalid signature! (Shared secret is incorrect.) I have updated to the cvs version of radiusclient-ng, but the problem still exists. It would appear that the md5 routine is not compatible with the md5 routine used by the freeradius software. Just wondering is anyone else has worked their way around this problem. Cheers, Stuart > X-Archived: http://article.gmane.org/gmane.comp.voip.openser.user/4415 > From: Daniel-Constantin Mierla <daniel@...> > Subject: FYI: bug in radiusclient ng for 64b > Newsgroups: gmane.comp.voip.openser.user, gmane.comp.voip.openser.devel > Date: 2006-05-16 01:44:16 GMT (13 weeks, 3 days and 58 minutes ago) > > On 05/16/06 02:48, Maxim Sobolev wrote: >> Hi, >> >> Reportedly there is an problem in radiusclient library with md5 hash >> calculation on 64bit platforms (amd64, sparc64, etc), which results in >> server rejecting requests even with proper secret configured on >> client. I have applied a fix, but don't have access to any 64bit >> hardware, therefore can't verify. >> >> Therefore, if anybody is having problems with radiusclient on 64bit >> arch he is more than welcome to try update to the latest version from >> cvs and let me know if it helps or not. Instructions on checking out >> sources from berlios cvs are here: >> https://developer.berlios.de/cvs/?group_id=1208. >> >> Thanks! >> >> -Maxim >

3 2

[Users] OpenSER administrator issue
by Pat wang 27 Feb '07

27 Feb '07

Hi, Just managed to install the new openser administrator and everything just works. Thanks to Mike for this nice application. However I can't add user with openserctl anymore. The following error appears: ERROR 1054 (42S22) at line 1: Unknown column 'phplib_id' in 'field list' ERROR: introducing the new user '5702' to the database failed On the other hand, adding new users from administrator GUI does not have this problem. This happened right after installation of aministrator so I think it is related to the administrator mysql table changes. Any suggestions please? Thanks, Patrick

4 4

[Users] eta for v1.2.0 - March 12
by Daniel-Constantin Mierla 27 Feb '07

27 Feb '07

Hello everybody, just to give you an update with estimation time line for release 1.2.0... The plan is to release it on the 12th of March, hard work being with packaging files update, migration tools, and of course still with testing and stress-testing ... If you have good DB (mysql/postgres) skills and want to help, just say it, it is enough work around :-) . Some tables changed the structure of the fields (like LCR with the IP address), and a bit more advanced migration mechanism should be in place. Of course, help with testing is always appreciated ... Cheers, Daniel

1 0

[Serusers] PATH Module in SER?
by SER LIST 27 Feb '07

27 Feb '07

Is there a paths module for SER? I want to implement a Load balancer using Dispatcher module and I will need a path module for handling registrations. Thanks in advance _________________________________________________________________ Your Space. Your Friends. Your Stories. Share your world with Windows Live Spaces. http://spaces.live.com/?mkt=en-ca

1 0

[Users] OpenSER CVS was migrated to SVN
by Daniel-Constantin Mierla 27 Feb '07

27 Feb '07

Hello, after the discussion on devel mailing list was decided to migrate OpenSER CVS repository to SVN. Most of developers considered the SVN more convenient to work with, especially when hunting reported bugs (svn revision number is printed by openser -V). Unfortunately this bring a bit of work for those installing and maintaining OpenSER from CVS. The have to grab the subversion (svn) tree righ now. There are no changes in directory structure, so any patch you had privately should apply easy. To get the same directory tree as for CVS: # svn co https://openser.svn.sourceforge.net/svnroot/openser sip-server For details about how to get a certain branch, please see the download page on http://www.openser.org: http://www.openser.org/index.php?option=com_content&task=view&id=29&Itemid=… We are still working on making snapshot system using SVN, and maybe some other tools are not yet updated, but they will be in the next days. However, what is most important should be ready to go -- source tree should be fully functional with SVN. Please let us know if you encounter any issue. The translation to SVN was made over this weeken. New commits were already done to SVN. There was no critical update you should run after, but is recommended to get SVN version as soon as possible. Old branches were migrated (releases 0.9.x, 1.0.x, 1.1.x). For the reference, initial revision for subversion is 1689. Once you get your patch against latest CVS version availabe, then should be no issue to apply it to this subversion revision. Full commit history from CVS is available on SVN. Hope SVN will make openser maintenance easier for all of you. Here are some links to get you familiar with SVN commands and capabilities: http://svnbook.red-bean.com/nightly/en/index.html http://subversion.tigris.org/ OpenSER SVN page at sourceforge: http://sourceforge.net/svn/?group_id=139143 Cheers, Daniel

4 6

Re: [Users] dialog module configuration question
by Bogdan-Andrei Iancu 26 Feb '07

26 Feb '07

Hi Andy, yes, you are correct - the package 7 (the 200 OK) must mirror the Record-Route set from the request. For this you need to enable the rrs param : <recv request="INVITE" crlf="true" rrs="true"> </recv> regards, bogdan Andy Pyles wrote: > Hi Bogdan, > > ok let me go back to my example: > > Here's more detail: > 192.168.0.101 = Caller (sipp uas) > 1.2.3.4 = openser > 4.3.2.1 = callee ( sipp uac) > > > 1.) 192.168.0.101 -> 1.2.3.4 SIP/SDP Request: INVITE > sip:service@1.2.3.4:5060, with session description > 2.) 1.2.3.4 -> 192.168.0.101 SIP Status: 100 Giving a try > 3.) 1.2.3.4 -> 4.3.2.1 SIP/SDP Request: INVITE > sip:service@4.3.2.1:5060, with session description > 4.) 4.3.2.1 -> 1.2.3.4 SIP Status: 180 Ringing > 5.) 4.3.2.1 -> 1.2.3.4 SIP/SDP Status: 200 OK, with session > description > 6.) 1.2.3.4 -> 192.168.0.101 SIP Status: 180 Ringing > 7.) 1.2.3.4 -> 192.168.0.101 SIP/SDP Status: 200 OK, with session > description > 8.) 192.168.0.101 -> 1.2.3.4 SIP Request: ACK > sip:service@1.2.3.4:5060 > 9.) 1.2.3.4 -> 4.3.2.1 SIP Request: ACK sip:service@4.3.2.1:5060 > 10.) 192.168.0.101 -> 1.2.3.4 SIP Request: BYE > sip:service@1.2.3.4:5060 > 11.) 1.2.3.4 -> 4.3.2.1 SIP Request: BYE sip:service@4.3.2.1:5060 > 12.) 4.3.2.1 -> 1.2.3.4 SIP Status: 200 OK > 13.) 1.2.3.4 -> 192.168.0.101 SIP Status: 200 OK > > So, you are saying for Packets 8, 10 I should add the '[routes]' logic > to sipp. How this works is: from the sipp documentation: "rrs: Record > Route Set. if this attribute is set to "true", then the > "Record-Route:" header of the message received is stored and can be > recalled using the [routes] keyword.". > > This I completey agree with. sipp Must be sending the Route: header > in Packets 8 and 10. However, packet 7 MUST have the Record-route > header, otherwise, How can sipp can put the correct value into the > Route: header. See my point? > > Reference: rfc 3665 ( secion 3.2 Packet f11, f14) > > regards, > Andy > > > On 2/23/07, Andy Pyles <andy.pyles(a)gmail.com> wrote: >> Hi Bogdan, >> >> correct. but on client config "[routes]" ( for sipp) will only work >> IF the client receives a Record-route. Since I'm not, it doesn't help >> me. Am I missing something? >> >> Andy >> >> On 2/23/07, Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> wrote: >> > Hi Andy, >> > >> > in client config, you need to add "[routes]" for ACK and BYE messages >> > (take a look at the cfg I sent you) >> > >> > regards, >> > bogdan >> > >> > Andy Pyles wrote: >> > > I Just re-read the docs on loose_route(). So please disregard this >> > > question. ( only processed if Route: header is present. Which isn't >> > > present because Record-route: header isn't being sent to caller ) >> > > >> > > So, I'm still trying to figure out why record-route: header is not >> > > being sent to caller. >> > > >> > > >> > > On 2/22/07, Andy Pyles <andy.pyles(a)gmail.com> wrote: >> > >> Hi Bogdan, >> > >> >> > >> After running additional debugs, for some reason the call to >> > >> loose_route() is failing. >> > >> >> > >> if (loose_route()) { >> > >> # mark routing logic in request >> > >> xlog("L_INFO", "loose_route() succeeded\n "); >> > >> route(1); >> > >> } else{ >> > >> xlog("L_INFO", "loose_route()failed - M=$rm RURI=$ru F=$fu >> > >> T=$tu IP=$si ID=$ci\n"); >> > >> }; >> > >> >> > >> >> > >> Any ideas why this could be occuring? >> > >> >> > >> >> > >> On 2/22/07, Andy Pyles <andy.pyles(a)gmail.com> wrote: >> > >> > HI Bogdan, >> > >> > >> > >> > I'm already using an almsot identical version of uas.xml and >> uac.xml ( >> > >> > yes rrs=true) is being used. However in your version the uas.xml >> > >> > doesn't have rrs="true" after initial invite which I think is >> needed. >> > >> > See as you can see below, setting rrs="true" for uac will only >> work if >> > >> > it receives a Record-Route header in the 200OK which it's not. >> > >> > >> > >> > In this case, ALL messages from openser to sipp uac do not >> contain the >> > >> > Record-route header. So I don't think it's a sipp problem, but an >> > >> > openser configuration problem. I've tried using other devices >> for a >> > >> > uac, such as x-lite but the same problem. >> > >> > >> > >> > Andy >> > >> > >> > >> > On 2/22/07, Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> wrote: >> > >> > > Hi Andy, >> > >> > > >> > >> > > so it's about sipp :D - I remember I had some hard times to >> make >> > >> it work >> > >> > > with record Route. >> > >> > > >> > >> > > take a look at the attached files, they might help you. >> > >> > > >> > >> > > regards, >> > >> > > bogdan >> > >> > > >> > >> > > Andy Pyles wrote: >> > >> > > > HI Bogdan, >> > >> > > > >> > >> > > > thanks for your reply. >> > >> > > > yes you are correct. The Bye doesn't have the Route header. >> > >> > > > It appears the the 200 OK sent to the caller doesn't >> contain a >> > >> > > > Record-route header. >> > >> > > > Messages between openser and callee contain record-route >> > >> information, >> > >> > > > but messages between caller and openser do not. >> > >> > > > Is there a way to enable that? >> > >> > > > >> > >> > > > Here's more detail: >> > >> > > > 192.168.0.101 = Caller (sipp) >> > >> > > > 1.2.3.4 = openser >> > >> > > > 4.3.2.1 = callee ( sipp) >> > >> > > > >> > >> > > > >> > >> > > > 1.) 192.168.0.101 -> 1.2.3.4 SIP/SDP Request: INVITE >> > >> > > > sip:service@1.2.3.4:5060, with session description >> > >> > > > 2.) 1.2.3.4 -> 192.168.0.101 SIP Status: 100 Giving a try >> > >> > > > 3.) 1.2.3.4 -> 4.3.2.1 SIP/SDP Request: INVITE >> > >> > > > sip:service@4.3.2.1:5060, with session description >> > >> > > > 4.) 4.3.2.1 -> 1.2.3.4 SIP Status: 180 Ringing >> > >> > > > 5.) 4.3.2.1 -> 1.2.3.4 SIP/SDP Status: 200 OK, with >> > >> session >> > >> > > > description >> > >> > > > 6.) 1.2.3.4 -> 192.168.0.101 SIP Status: 180 Ringing >> > >> > > > 7.) 1.2.3.4 -> 192.168.0.101 SIP/SDP Status: 200 OK, with >> > >> session >> > >> > > > description >> > >> > > > 8.) 192.168.0.101 -> 1.2.3.4 SIP Request: ACK >> > >> > > > sip:service@1.2.3.4:5060 >> > >> > > > 9.) 1.2.3.4 -> 4.3.2.1 SIP Request: ACK >> > >> sip:service@4.3.2.1:5060 >> > >> > > > 10.) 192.168.0.101 -> 1.2.3.4 SIP Request: BYE >> > >> > > > sip:service@1.2.3.4:5060 >> > >> > > > 11.) 1.2.3.4 -> 4.3.2.1 SIP Request: BYE >> > >> sip:service@4.3.2.1:5060 >> > >> > > > 12.) 4.3.2.1 -> 1.2.3.4 SIP Status: 200 OK >> > >> > > > 13.) 1.2.3.4 -> 192.168.0.101 SIP Status: 200 OK >> > >> > > > >> > >> > > > --- >> > >> > > > Packets 6,7 and following contain no Record-route >> information. >> > >> > > > The other weird thing is that openser is passing on the >> Route: >> > >> header >> > >> > > > it recevied from callee to the caller. >> > >> > > > >> > >> > > > >> > >> > > > Please see attached for complete ngrep output. >> > >> > > > >> > >> > > > >> > >> > > > On 2/21/07, Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> >> wrote: >> > >> > > >> Hi Andy, >> > >> > > >> >> > >> > > >> could you check on the net if the BYE contain the Route hdr >> > >> added to >> > >> > > >> INVITE as Record-Route? I have some doubts on this as I see: >> > >> > > >> 0(966) find_first_route: No Route headers found >> > >> > > >> 0(966) loose_route: There is no Route HF >> > >> > > >> >> > >> > > >> and if the BYE is not identified, the dialog is not closed. >> > >> > > >> >> > >> > > >> regards, >> > >> > > >> bogdan >> > >> > > >> >> > >> > > >> Andy Pyles wrote: >> > >> > > >> > Hello, >> > >> > > >> > >> > >> > > >> > I have a question on how to configure the dialog module ( >> > >> 1.2.x from >> > >> > > >> > cvs yesterday ). >> > >> > > >> > >> > >> > > >> > With my config, ( attached) I can make calls and have >> > >> verified that >> > >> > > >> > the acc module is working correctly. >> > >> > > >> > >> > >> > > >> > My question is, when I enable the dialog module, I can see >> > >> that it is >> > >> > > >> > incrementing call count correctly, but when a bye is >> > >> received, the >> > >> > > >> > dialog:active_dialogs statistic is never decremented. >> > >> > > >> > >> > >> > > >> > In the debug level 9 logs, ( also attached) I see this >> error >> > >> after the >> > >> > > >> > 200OK is sent to the bye: >> > >> > > >> > >> > >> > > >> > 1(969) DBUG:dialog:unref_dlg: unref dlg 0xa7ce5a98 with 1 >> > >> > > >> (delete=0)-> 1 >> > >> > > >> > >> > >> > > >> > Is this a case of one of the timers being set too >> short? by >> > >> the way >> > >> > > >> > using a variable call length from well under a second ( >> > >> using sipp ) >> > >> > > >> > to 20 second call doesnt' seem to make a difference . >> > >> > > >> > >> > >> > > >> > >> > >> > > >> > Thanks, >> > >> > > >> > Andy >> > >> > > >> > >> > >> >

2 3

[Users] Exported statistics and pseudo-variables
by Michel Bensoussan 26 Feb '07

26 Feb '07

Hello I didn't understand how to use the exported statistics and the pseudo-variables. In module dialog, active_dialogs is an exported statistic and $dlg_count a pseudo variable. How can I get them in code and in cfg file? Can I write in cfg file something like: if ($dlg_count > 10){...}? (not in this way of course but that's the idea). Thanks, Michel. PS: I didn't understand the example in http://www.openser.org/dokuwiki/doku.php/pseudovariables:1.1.x -- Michel Bensoussan Extricom Ltd. Tel-Aviv University, The Wolfson Computer and Software Engineering Building 30 Haim Levanon Street, P.O.B. 39040 Tel-Aviv, 61390 Israel Tel: +972 3 745 1000 ext 1065 Fax: +972 3 745 1001

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[Users] Adding openser to an already in use asterisk pbx
by Ben Ferguson 26 Feb '07

26 Feb '07

Hello all, I am very new to openSER and have been going over MANY, MANY pages of info on openSER (and SER) and am having a bit of a problem figuring out what exactly i need to do for what I am trying to have happen. Basically it's like this: I have a small company running a NATed asterisk pbx (there is actually about 30 to 40 users already setup and using Asterisk). Now I would like to add openSER to allow some outsourced/remote users via voip so they may dial anyone in the company directly over their voip connection. Basically, all asterisk users can call all asterisk users and also call remote (openSER) users; and all remote (openSER) users can call Asterisk users. I know Asterisk decently well, but I am not sure what I would need to set up in openSER to handle this. My Asterisk box is currently not set up for DB auth. I just have the users set up in sip.conf. Re-doing the Asterisk box is absolutely out of the scope of this project. I would just like to set up some remote users via openSER. (I do have a public IP set up for my openSER box.) Anyone have this type of setup going? Or anyone can tell me how this would best be done? (I'd love to see an openser.cfg of this setup....) Thanks in advance, Ben Jeffries fjsixty [at] gmail [dot] com

1 0

[Users] TLS and dead tcp child
by Paulo Angonese 26 Feb '07

26 Feb '07

Hi, I have the same problem. When the SNOM ( 360 SIP 6.5.2 ) sends the REGISTER using TLS OpenSER crashes. I´m using OpenSer (1.1.1-tls (i386/linux)), openssl-0.9.7f-7.10 openssl-devel-0.9.7f-7.10 Fedora 2.6.11-1.1369_FC4 Any help? Thanks Paulo Angonese [Users] CVS 31.03, TLS and dead tcp child Bogdan-Andrei Iancu bogdan at voice-system.ro Tue Oct 31 17:13:43 CET 2006 * Previous message: [Users] CVS 31.03, TLS and dead tcp child * Next message: [Users] Q:ABOUTE:SET UP MESSAGE FLOW * Messages sorted by: [ date ] [ thread ] [ subject ] [ author ] Hi Frank, can you pack the cores + binaries (lib + exe) + sources+logs(in debug mode) and put it somewhere for download? I will try to take a look. regards, Bogdan Frank DInnocenzo wrote: > Bogdan, > > I am using openser-1.1.0-tls_src. I actually get two core files after > the crash. Can I post them somehow? > > > # openser -V > version: openser 1.1.0-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. > @(#) $Id: main.c,v 1.20 2006/07/04 17:25:54 bogdan_iancu Exp $ > main.c compiled on 14:27:50 Oct 24 2006 with gcc 4.0.0 > > Thanks > Frank > > > */Bogdan-Andrei Iancu <bogdan at voice-system.ro>/* wrote: > > Hi Frank, > > I'm successfully using a SNOM 320 with TLS and no problem so far..... > What openser version are you using?...do you get any core file > after the > crash?? > > regards, > bogdan > > Frank DInnocenzo wrote: > > > Hello, > > > > I'm new to openSer and wondered if there was any resolution to this > > post??? I'm hitting this same crash with openSer (with TLS enabled) > > using a Snom phone (300 with latest Beta firmware - > > snom300-6.5-SIP-j.bin ). I've gone > > through the archives looking for help getting this setup > working. I've > > also tried the Snom Softphone with the same result. > > > > Thanks > > Frank > > > > >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>> > > > > Hi, > > > > I have tried to register to OpenSER CVS 31.03 with my SNOM 360 > > (5.3.6sw), disabled pretty all modules from OpenSER (radius, > xlog, auth, > > ...) and always the server crashes. > > > > Is there something I have missed in configuration ? > > > > Thanks a lot for help, > > -Mika > > > > The error message and debug info seems like this: > > > > 11(24351) tcpconn_new: new tcp connection to: XXX.XX.XX.XXX > > 11(24351) tcpconn_new: on port 2171, type 3 > > 11(24351) tls_tcpconn_init: Entered: Creating a whole new ssl > connection > > 11(24351) tls_tcpconn_init: Looking up tls domain > [XXX.XX.XX.XXX:5061] > > 11(24351) tls_tcpconn_init: Using default tls server settings > > 11(24351) tls_tcpconn_init: Setting in ACCEPT mode (server) > > 11(24351) tcpconn_add: hashes: 442, 1 > > 11(24351) handle_new_connect: new connection: 0xb608f6a0 24 > flags: 0002 > > 11(24351) send2child: to tcp child 0 7(24347), 0xb608f6a0 > > 7(24347) received n=4 con=0xb608f6a0, fd=19 > > 7(24347) DBG: io_watch_add(0x810e580, 19, 2, 0xb608f6a0), fd_no=1 > > 7(24347) tls_update_fd: New fd is 19 > > 11(24351) DBG: handle_tcp_child: dead tcp child 0 (pid 24347, no 7) > > (shutting down?) > > 11(24351) DBG: io_watch_del (0x810e420, 17, -1, 0x0) fd_no=16 called > > 11(24351) ERROR: receive_fd: EOF on 15 > > 11(24351) DBG: handle_ser_child: dead child 7, pid 24347 > (shutting down?) > > 11(24351) DBG: io_watch_del (0x810e420, 15, -1, 0x0) fd_no=15 called > > 0(24339) child process 24347 exited by a signal 11 > > 0(24339) core was generated > > 0(24339) INFO: terminating due to SIGCHLD > > 6(24346) INFO: signal 15 received > > 6(24346) Memory status (pkg):

2 3

[Users] FOSDEM 2007 - openser slides
by Daniel-Constantin Mierla 26 Feb '07

26 Feb '07

Hello, if anybody interested in the slides of openser simple-xmpp gateway talk at FOSDEM 2007: http://www.openser.org/events/2007-FOSDEM/slides/fosdem-2007_daniel-constan… Some details about the event: http://www.openser.org/index.php?option=com_content&task=view&id=93&Itemid=9 Cheers, Daniel

1 0

[Users] Registration of Polycom SoundPointIP phone with OpenSER
by Sheetal Bhogale 26 Feb '07

26 Feb '07

Hello all, I am trying to get this Polycom phone to register with TLS but with no luck. Can anyone help me with what certificate are you guys using,is it ur own or the defalut certificate.This phone doesn't seem to be liking my certificate netiher its able to register with the default certificate. Also anything with the firmware since I am not able to get this phone registered as well. Any help would be appreciated. Thanks, Sheetal

1 0

[Serusers] Strange Via parsing error....
by sip 26 Feb '07

26 Feb '07

I'm getting an odd Via parsing error from SER 0.9.6: Feb 23 16:52:49 death ser[17389]: error: parse_via_param Feb 23 16:52:49 death ser[17389]: ERROR: parse_via on: <sip/2.0/udp 172.30.237.149:56755;branch=z9hg4bk-d87543-d75bc86d826ac929-1--d87543-> Feb 23 16:52:49 death ser[17389]: ERROR: parse_via parse error, parsed so far:<sip/2.0/udp 172.30.237.149:56755;branch=z9hg4bk-d87543-d75bc86d826ac929-1--d87543-> Feb 23 16:52:49 death ser[17389]: ERROR: get_hdr_field: bad via Feb 23 16:52:49 death ser[17389]: ERROR: parse_msg: message=<REGISTEl sip:proxy.ideasip.com SIP/2.0> Feb 23 16:52:49 death ser[17389]: ERROR: receive_msg: parse_msg failed When I look at the packet, it looks like the actual SIP data is somehow getting garbled... with odd characters showing up in the middle of headers, etc. Any idea what might cause this? U 148.233.151.30:43764 -> XX.XX.XX.XX:5060 REGISTEl sip:proxy.ideasip.com SIP/2.0. Via: sip/2.0/udp 172.30.237.149:16240;branch=z9hg4bk-d87543-9332b73b5700e95c-1--d87543-. Max-Forward2a:70. Contactm:<sip:user@148.233.151.30:32332;rinstance=6c0c8351d99e79db>. To: "mario"<sip:user@proxy.ideasip.com>. From: "mario"<sip:user@proxy.ideasip.com>;tag=fe132761. Call-ID: n2m5owi1odc4mzkznmm5mjflmzvmzmu3zgjmngqym2y.. CSe1h:1 register. Expire1k:3600. Allo0b:invite, ack, cancel, options, bye, refer, notify, message, subscribe, info. User-Agen5m:x-lite release 1006e stamp 34025. Content-Lengthl:0. As you can see... things like REGISTEl, Max Forward2a: Expire1k: All these things look garbled. Would this be a transmission error of some kind (the garbled headers are identical for each submitted packet, though, so it seems unlikely) ? N.

4 3

[Users] failover routing and LCR
by Howard Tang 26 Feb '07

26 Feb '07

Hi, I would like to set up LCR and failover routing in my openser. I want to find some working examples on how i can set that up. Can someone point me to the right direction please? If any one can offer me a sample config that will be great. -- Howard Tang

2 1

[Users] Modules Index Updated
by Edson 26 Feb '07

26 Feb '07

I'm glad to inform all the members of the list that ALL modules, from ALL OpenSER versions (0.9.x, 1.0.x, 1.1.x and 1.2.x) have been updated and included on the Wiki page (http://www.openser.org/dokuwiki/doku.php/start). All index are presented in a table format with three columns: the first has the name of the 'thing' being referenced; the second the module from where that reference came from; and the third is for user comments. All indexes were generated after the files on the web, so it's possible that there are some inconsistences between this files and the ones on CVS. Please let me know about any update, so that I can re-run the script to update the indexes. Due to automatization, some documentation problems pops up. like the use of a '$' sign in front of a MI command on the dialog module. They are others. what I found I corrected manually.. So, feel free to use it, and don't forget to re-inform the list about any changes on the original files. Sugestions, as always, are welcome. ;) Edson.

2 1

Fwd: Re: [Serusers] xlite and ser
by zahra bahar 26 Feb '07

26 Feb '07

Note: forwarded message attached. ____________________________________________________________________________________ We won't tell. Get more on shows you hate to love (and love to hate): Yahoo! TV's Guilty Pleasures list. http://tv.yahoo.com/collections/265

1 0

[Serusers] Problem with SER
by Jesús Terán 26 Feb '07

26 Feb '07

Hello, i am using the ser 0.8.12 and i am getting the next error in the logs: Feb 22 09:47:13 117 /usr/sbin/ser[11779]: ERROR: warning_builder: buffer size exceeded Feb 22 09:47:13 117 /usr/sbin/ser[11779]: WARNING: warning skipped -- too big Feb 22 09:47:16 117 /usr/sbin/ser[11779]: ERROR: warning_builder: buffer size exceeded Feb 22 09:47:16 117 /usr/sbin/ser[11779]: WARNING: warning skipped -- too big And when this happen some process block and dont permit to make and to do calls. Can anyone help me? My ser is on a Red Hat enterprise linux 4

2 3

RE: [Users] cancelling INVITE during OK response
by Web Monkey 26 Feb '07

26 Feb '07

hi, thanks for your response. is the sending of CANCEL handled by the openser engine internally? i need to do this from the openser config file. basically, my statement is like this: onreply_route[2] { if (status =~ "(183)|(2[0-9][0-9])") { if (client_nat_test("1")) { fix_contact(); append_hf("P-hint: Reply applied\r\n"); } use_media_proxy(); } if (status == "200") { invoke_my_function_from_here("$fu:$tu:$ci"); if (retcode == 1) { # propagate OK to caller; do nothing } else { # terminate or cancel INVITE from here } } } i should also be able to do this in proxy1 from the figure below. i think i need to send CANCEL to UAC1 and BYE to UAC2. this is my understanding of the RFC. someone may have implemented a function or module to handle this. thanks again. "Premnath, KN" <Premnath.KN(a)siemens.com> wrote: Hi Sho, When you get 200OK, you send an "ACK", so it is a good indication for the initial "INVITE" that it is ACKnowledged. When you get an error, anyway you will not send "ACK" instead still send the "CANCEL" so the UAC2 is clearly handled (and it will not wait for ACK), give a try !!! Thank You,. With warm regards, Premnath Narayanan --------------------------------- From: users-bounces(a)openser.org [mailto:users-bounces@openser.org] On Behalf Of Web Monkey Sent: Monday, February 26, 2007 9:24 AM To: users(a)openser.org; users(a)iptel.org Subject: [Users] cancelling INVITE during OK response hi, is there a way to cancel or terminate an INVITE or call upon receipt of the OK reply. my configuration invokes an external program which either return success or failure for OK message (response to INVITE). when failure is returned, i want the script to terminate the call or cancel the INVITE (OK message will not be forwarded to the calling UA). is there an existing function or module to realize this? the scenario is drawn below with provisional messages removed. UAC1 ------------------ proxy1 ----------------- proxy2 --------------- UAC2 call: INVITE ---------------------------> INVITE ----------------------> INVITE -----------------> OK <---------------- *XXX - error occurred during processing OK message thanks, sho --------------------------------- It's here! Your new message! Get new email alerts with the free Yahoo! Toolbar. The new company Nokia Siemens Networks is expected to start operations in the first quarter of 2007, subject to customary regulatory approvals, the completion of standard closing conditions, and the agreement of a number of detailed implementation steps. Disclaimer: This e-mail may contain trade secrets or privileged, undisclosed or otherwise confidential information. If you have received this e-mail in error, you are hereby notified that any review, copying or distribution of it is strictly prohibited. Please inform us immediately and destroy the original transmittal. Thank you for your cooperation. --------------------------------- Check out the all-new Yahoo! Mail beta - Fire up a more powerful email and get things done faster.

1 0

[Serusers] SER Users and Developers Meeting in Prague, March, colocated with the IETF
by Jiri Kuthan 25 Feb '07

25 Feb '07

Ladies and gentlemen, I'm glad to invite you to a meeting of SER users and developers in March 2007, in Prague, Czech Republic. See more at http://www.iptel.org/ietf_meeting I'm looking forward to seeing you there. In the meantime, please register yourself if you plan to attend and don't hesitate to make any agenda suggestions -- SER related ideally, even though any "social" suggestions are perfectly fine too. I can promise you it will be a wonderful event -- Prague is beatiful, SER is beatiful, and so is the folks involved in SER development -- LET'S MEET! -jiri -- Jiri Kuthan http://iptel.org/~jiri/

4 4

[Users] mediaproxy exceptions on session delete
by Klaus Darilion 25 Feb '07

25 Feb '07

Hi! I have a problem with mediaproxy 1.8. Whenever a session is deleted there is an exception: Feb 14 14:18:27 foobar mediaproxy[27053]: delete OGI3Nzk1N2ExODY4NTIzYzIxMjc1NmYxN2MwMDBiZjU. info= Feb 14 14:18:27 foobar mediaproxy[27053]: session OGI3Nzk1N2ExODY4NTIzYzIxMjc1NmYxN2MwMDBiZjU.: 10558/10557/21113 packets, 778927/774750/1553531 bytes (caller/called/relayed) Feb 14 14:18:27 foobar mediaproxy[27053]: session OGI3Nzk1N2ExODY4NTIzYzIxMjc1NmYxN2MwMDBiZjU.: ended. Feb 14 14:18:27 foobar mediaproxy[27053]: error: uncaptured python exception, closing channel <rtphandler.CommandHandler connected 5.6.7.8:38879 at 0xb7d255ec> (exceptions.NameError:global name 'StopRecordSerializer' is not defined [/usr/lib/python2.3/asyncore.py|readwrite|86] [/usr/lib/python2.3/asyncore.py|handle_read_event|390] [/opt/mediaproxy/modules/rtphandler.py|handle_read|441] [/opt/mediaproxy/modules/rtphandler.py|process|568]) Feb 14 14:19:53 foobar mediaproxy[27053]: status Feb 14 14:19:53 foobar mediaproxy[27053]: execution time: 0.11 ms I could not find the problem - accounting.py with StopRecordSerializer is there, and it is imported in rtphandler. Do others also have problems and maybe a workaround? regards klaus -- Klaus Darilion nic.at

2 1

Fwd: [Serusers] xlite and ser
by zahra bahar 25 Feb '07

25 Feb '07

Note: forwarded message attached. ____________________________________________________________________________________ Food fight? Enjoy some healthy debate in the Yahoo! Answers Food & Drink Q&A. http://answers.yahoo.com/dir/?link=list&sid=396545367

2 1

[Serusers] reminder: PRague meeting registration deadline approaching
by Jiri Kuthan 24 Feb '07

24 Feb '07

dont forget to register yourself if you plan to attend -- we really need to dimension the meeting facilities properly. http://www.iptel.org/ietf_meeting/participants Looking forward to seeing ya all! -jiri -- Jiri Kuthan http://iptel.org/~jiri/

1 0

[Users] new CDRTool release 5.0.10
by Adrian Georgescu 24 Feb '07

24 Feb '07

Hello, CDRTool 5.0.10 has been released. The software can be downloaded from: http://download.dns-hosting.info/CDRTool/ Changelog from 5.0.5: cdrtool (5.0.10) unstable; urgency=low * Fixed reload of prepaid accounts from rating page to reload only the prepaid account that has changed (when PHP register_globals if off) cdrtool (5.0.9) unstable; urgency=low * Added /var/www/CDRTool/scripts/replicationStatus.php script to easily check the database replication process Add to global.inc $CDRTool['replicated_databases']=array ('DB1','DB2'); and defined DB1 and DB2 connections to the mysql databases that replicate to each other. The connections must use the same IP addresses and usernames used during the setup of the replication process cdrtool (5.0.8) unstable; urgency=low * Enable failover between CDR databases for normalization process db_class can be an array with database connection classes * Removed db_class_readonly from global.inc cdrtool (5.0.7) unstable; urgency=low * Fixed version to be compliant with debian native version numbering cdrtool (5.0-6) unstable; urgency=low * Avoid unnecessary sql OR clause in CDR search that caused slow queries Regards, Adrian Georgescu

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Re: [Serusers] Call from Asterisk to SER to a foreign subscriber
by karim basraoui 24 Feb '07

24 Feb '07

i had : UA1 (192.168.4.66:5061) }------------->Asterisk----------->SER-----------> UA3 (192.168.4.65:5061) UA2 (192.168.4.66:5062) then UA1 moved to SER domain : UA3 (192.168.4.65:5061) UA2 (192.168.4.66:5062) ------------->Asterisk----------->SER-----------> UA1 (192.168.4.66:5061) UA1 registers to SER and SER forwards the registration to ASTERISK, Asterisk is aware of the new location. UA2 tries to call UA3. It sends the invite to Asterisk and Asterisk replies with a TRYING message. after a while the call fails with an error 2007/2/24, ram <talk2ram(a)gmail.com>: > > then register with SER, should do the job > > UA----ASterisk----SER-----Phone > > is this correct > > ram > > > On 2/24/07, karim basraoui <basraouik(a)gmail.com> wrote: > > > > Hi, > > > > the call i'm trying to establish is from ASTERISK to SER. > > the UA i'm calling was initially in ASTERISK domain and moved to SER > > domain , a roaming case. > > > > 2007/2/24, ram < talk2ram(a)gmail.com>: > > > > > > Hi > > > > > > your question is not clear > > > > > > how is the call path happening from ASterisk or from SER > > > or both are gateway proxies ? > > > > > > ram > > > > > > > > > On 2/23/07, karim basraoui < basraouik(a)gmail.com > wrote: > > > > > > > Hi, > > > > > > > > I've set up a small architecture in which an ASterisk proxy is > > > > connected to a SER proxy in two different domains. > > > > > > > > I have sip phones connected to both of them. when an asterisk > > > > subscriber "travels" to SER domain it tries to register to SER. I've > > > > configured SER to accept foreign > > > > subscribers and to REGISTER them to their initial domain by the > > > > command "t_replicate". Till now everything works good, the UA is registered > > > > and Asterisk is aware > > > > of its new location, it displays in peers list the UA and its new IP > > > > address and port. > > > > > > > > The problem is when i'm trying to call the foreign UA from an > > > > Asterisk subscriber, it doesn't work. i have a TRYING response but no call > > > > establishement. > > > > > > > > how can i overcome this problem please? > > > > > > > > -- > > > > BASRAOUI Karim > > > > www.basraouik.afrikart.net > > > > _______________________________________________ > > > > Serusers mailing list > > > > Serusers(a)lists.iptel.org > > > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > > > > > > > > > > > > > -- > > BASRAOUI Karim > > www.basraouik.afrikart.net > > > > -- BASRAOUI Karim www.basraouik.afrikart.net

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Re: [Serusers] Call from Asterisk to SER to a foreign subscriber
by karim basraoui 24 Feb '07

24 Feb '07

Hi, the call i'm trying to establish is from ASTERISK to SER. the UA i'm calling was initially in ASTERISK domain and moved to SER domain , a roaming case. 2007/2/24, ram <talk2ram(a)gmail.com>: > > Hi > > your question is not clear > > how is the call path happening from ASterisk or from SER > or both are gateway proxies ? > > ram > > > On 2/23/07, karim basraoui <basraouik(a)gmail.com> wrote: > > > Hi, > > > > I've set up a small architecture in which an ASterisk proxy is connected > > to a SER proxy in two different domains. > > > > I have sip phones connected to both of them. when an asterisk subscriber > > "travels" to SER domain it tries to register to SER. I've configured SER to > > accept foreign > > subscribers and to REGISTER them to their initial domain by the command > > "t_replicate". Till now everything works good, the UA is registered and > > Asterisk is aware > > of its new location, it displays in peers list the UA and its new IP > > address and port. > > > > The problem is when i'm trying to call the foreign UA from an Asterisk > > subscriber, it doesn't work. i have a TRYING response but no call > > establishement. > > > > how can i overcome this problem please? > > > > -- > > BASRAOUI Karim > > www.basraouik.afrikart.net > > _______________________________________________ > > Serusers mailing list > > Serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > -- BASRAOUI Karim www.basraouik.afrikart.net

1 0

[Serusers] iptel.org ANNOUNCEMENT: New SER bug marshal
by Greger V. Teigre 24 Feb '07

24 Feb '07

Hi all! I'm pleased to announce that Lakmal Silva has volunteered to be our SER bug marshal! Lakmal's main task will be to interface with the community and developers and make sure that bug fixes and feature requests don't disappear on the serusers and serdev mailing lists, as well as get them properly described/documented. (Please continue to use the tracker or mailing lists as you are used to.) The bug marshal's responsibilities are in more details: 1. Scan the mailings lists for bug reports, patches, feature requests, and improvement suggestions and: a) make sure they are registered on either http://iptel.org/ser/wishlist or as a tracker item (unless a developer picks it up) with unscheduled status b) request more information/debug traces/patches/etc in order to make the item possible to act upon for a developer c) solicit input from the developer(s) if clarification is needed 2. Pick up discussions on the mailing lists that concern new features, put new features on the wish list and if appropriate create a new specification/discussion page that will specify the feature in more detail and can be used for discussion purposes (examples: http://www.iptel.org/ser/specifications) Please don't contact Lakmal in private emails, but use the mailing lists if you have questions/issues. On behalf of the SER community, I welcome Lakmal and wish him luck in his new role as SER bug marshal! Greger

4 7

[Serusers] SER 0.9.6 with H.264 Problem!
by Steven Wong 24 Feb '07

24 Feb '07

Hello All: You will find enclosed our ser.cfg file that I have adapted from a very generous SIP user from the Internet. This ser.cfg file work perfectly with Audio (IP Phone and Softphone). But when I am trying to use a Video Phone (we did try with GrandStream GXV-3000 Video Phone), it just giving us Audio but NOT Video. I did also try the same GrandStream GXV-3000 Video Phone with FreeWorldDialup and it is working fine with Video. We have checked the etherreal output while trying to make a call and we are having this type of errors on the etherreal: ----- RTP Protocol Source IP (The IP of the Video Phone) and Destination IP (The IP of the SER Server) ------ Payload Type = Unknown (126) ICMP Protocol Source IP (The IP of the SER Server) and Destination IP (The IP of the Video Phone) ------Destination Unreachable (Port Unreacheable) These 2 errors repeated six time before it stop with a good signal of: Payload Type = ITU-T G.711 PCMU Then the audio start but without audio. --- Would you please tell us what is going wrong with this ser.cfg file? your answer is greatly appreciated. Many thanks, Steven Wong -------------------------------------------------------------------------- debug=3 fork=yes log_stderror=no check_via=no dns=no rev_dns=no listen=XXX.XXX.XXX.XXX # INSERT YOUR IP ADDRESS HERE # port=5060 children=3 alias="HOST.DOMAINNAME.COM" alias="XXX.XXX.XXX.XXX" fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:PASSWORD@localhost/ser" 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/nathelper.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/acc.so" modparam("auth_db|uri_db|usrloc", "db_url", "mysql://ser:PASSWORD@localhost/ser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") modparam("usrloc", "db_mode", 2) modparam("registrar", "nat_flag", 6) modparam("rr", "enable_full_lr", 1) #---Accounting params modparam("acc", "log_level", 1) modparam("acc", "log_flag", 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 >= max_len ) { sl_send_reply("513", "Message too big"); break; }; # !! Nathelper # Special handling for NATed clients; first, NAT test is # executed: it looks for via!=received and RFC1918 addresses # in Contact (may fail if line-folding is used); also, # the received test should, if completed, should check all # vias for rpesence of received if (nat_uac_test("3")) { # Allow RR-ed requests, as these may indicate that # a NAT-enabled proxy takes care of it; unless it is # a REGISTER if (method == "REGISTER" || ! search("^Record-Route:")) { log("LOG: Someone trying to register from private IP, rewriting\n"); # This will work only for user agents that support symmetric # communication. We tested quite many of them and majority is # smart enough to be symmetric. In some phones it takes a configuration # option. With Cisco 7960, it is called NAT_Enable=Yes, with kphone it is # called "symmetric media" and "symmetric signalling". fix_nated_contact(); # Rewrite contact with source IP of signalling if (method == "INVITE") { fix_nated_sdp("1"); # Add direction=active to SDP }; force_rport(); # Add rport parameter to topmost Via setflag(6); # Mark as NATed }; }; # 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; }; setflag(1); 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("HOST.DOMAINNAME.COM", "subscriber")) { www_challenge("HOST.DOMAINNAME.COM", "0"); break; }; save("location"); 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] { # !! Nathelper if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)" && !search("^Route:")){ sl_send_reply("479", "We don't forward to private IP addresses"); break; }; # if client or server know to be behind a NAT, enable relay if (isflagset(6)) { force_rtp_proxy(); }; # NAT processing of replies; apply to all transactions (for example, # re-INVITEs from public to private UA are hard to identify as # NATed at the moment of request processing); look at replies t_on_reply("1"); # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; } # !! Nathelper onreply_route[1] { # NATed transaction ? if (isflagset(6) && status =~ "(183)|2[0-9][0-9]") { fix_nated_contact(); force_rtp_proxy(); # otherwise, is it a transaction behind a NAT and we did not # know at time of request processing ? (RFC1918 contacts) } else if (nat_uac_test("1")) { fix_nated_contact(); }; } -------------------------------------------------------------------------- --------------------------------- The fish are biting. Get more visitors on your site using Yahoo! Search Marketing.

3 4

[Serusers] How to start 2 SER at once?
by Meidiana Ten 23 Feb '07

23 Feb '07

Hi there, I have problem in starting 2 SER at one machine at a time Can you guys show me how to start it.. I make 1 SER for home proxy (ser.cfg) and the other for SEMS (ser_sems.cfg) on same folder then I start ser.cfg with command #ser, it works well.. but when I try to start ser_sems.cfg with command #ser_sems, it doesn't work. It said it's unknown command Please show me where do I get wrong Thanx Regards, Meidiana ____________________________________________________________________________________ Need Mail bonding? Go to the Yahoo! Mail Q&A for great tips from Yahoo! Answers users. http://answers.yahoo.com/dir/?link=list&sid=396546091

3 2

Re: [Serusers] lookup_contacts() error after sd_lookup() from the speeddial module- latest cvs ottendorf
by TZieleniewski 23 Feb '07

23 Feb '07

2 5

[Users] ACK transaction
by Cseke Tamas 23 Feb '07

23 Feb '07

Hello, I'd like to relay sip calls from openserto another sip proxy (asterisk), and i have a problem with radius accounting, i 'd like to report acks, but openser doesn't send accounting packet on it. As i understand ack requests must match a transaction flaged to accounting. Config look like this, so i think all transaction is flaged. modparam("acc", "report_ack", 1) modparam("acc", "radius_flag", 1) modparam("acc", "radius_missed_flag", 2) route{ ... setflag(1); setflag(2); ... } Asterisk retransmit the 200 ok, so i think there is something mistake in the ack. as i read in rfc 3261, ACK must match the dialog, callid, from, to, fromtag, to, tag, and sequence number is relevant. and ack on a 200 ok, is always a new transaction, however i see branch parameter is 0 in the ack, it must start with the magic cookie z9hG4..., isn't it? I dont understand what could be the problem, why 200 ok is retransmitted, and why not openser send acctpacket to radius. Please someone take a look at the sip messages openser->asterisk INVITE sip:0619997400@127.0.0.1:5061 SIP/2.0 Record-Route: <sip:217.116.32.20;lr=on;ftag=LAWYzDCZnyMjZxIp> Via: SIP/2.0/UDP 217.116.32.20;branch=z9hG4bKde6a.a93de1c7.0 Via: SIP/2.0/UDP 192.168.1.150:1720;received=217.116.36.22;branch=z9hG4bKvDD2OX3SfnAE6cR5;rport=1720 Max-Forwards: 69 User-Agent: PA168S V1.53.006 CFG0 From: "222" <sip:222@siptest>;tag=LAWYzDCZnyMjZxIp To: "0619997400" <sip:0619997400@siptest> Call-ID: x3Gd6ZKUB9IeRuPO(a)192.168.1.150 Contact: <sip:222@217.116.36.2:1720> CSeq: 2 INVITE Supported: replaces Content-Type: application/sdp Content-Length: 275 [sdp] asterisk->openser SIP/2.0 200 OK Via: SIP/2.0/UDP 217.116.32.20;branch=z9hG4bKde6a.a93de1c7.0;received=217.116.32.20 Via: SIP/2.0/UDP 192.168.1.150:1720;received=217.116.36.2;branch=z9hG4bKvDD2OX3SfnAE6cR5;rport=1720 Record-Route: <sip:217.116.32.20;lr=on;ftag=LAWYzDCZnyMjZxIp> From: "222" <sip:222@siptest>;tag=LAWYzDCZnyMjZxIp To: "0619997400" <sip:0619997400@siptest>;tag=as37b57c9f Call-ID: x3Gd6ZKUB9IeRuPO(a)192.168.1.150 CSeq: 2 INVITE User-Agent: eWorld Com Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:0619997400@217.116.32.20:5061> Content-Type: application/sdp Content-Length: 206 [sdp] openser->asterisk <-- SIP read from 217.116.32.20:5060: ACK sip:0619997400@127.0.0.1:5061 SIP/2.0 Record-Route: <sip:217.116.32.20;lr=on;ftag=LAWYzDCZnyMjZxIp> Via: SIP/2.0/UDP 217.116.32.20;branch=0 Via: SIP/2.0/UDP 192.168.1.150:1720;received=217.116.36.2;branch=z9hG4bKKysauOpwGdpnsmlS;rport=1720 Route: <sip:217.116.32.20;lr=on;ftag=LAWYzDCZnyMjZxIp> Max-Forwards: 69 User-Agent: PA168S V1.53.006 CFG0 From: "222" <sip:222@siptest>;tag=LAWYzDCZnyMjZxIp To: "0619997400" <sip:0619997400@siptest>;tag=as37b57c9f Call-ID: x3Gd6ZKUB9IeRuPO(a)192.168.1.150 Contact: <sip:222@217.116.36.2:1720> Authorization: Digest username="222", realm="siptest", nonce="45dea111f92e1d1f9bf408a3996413168c304ee9", uri="sip:0619997400@217.116.32.20:5061", response="50419464515ca2771ec4b2bde7147219", algorithm=MD5 CSeq: 2 ACK Content-Length: 0 Thanks any help, Tamas

2 1

[Users] Authenticate INVITEs
by Octavarium 23 Feb '07

23 Feb '07

I have enabled mysql authentication of REGISTER messagess, but i need that only INVITE mesagges from devices properly registered in OpenSER can go through and make calls, and that INVITE mesagges from devices NOT registered are dropped. How can i do this? do you have a sample script? Thanx a lot!

2 1

[Users] Error Handling with Parallel Forking
by Chris Heiser 23 Feb '07

23 Feb '07

Hello gurus, I'm attempting to do some rather crazy parallel forking and have found a few issues I'm hoping to solve. I thought I'd solicit some advice from the experts. Some background: Each device registers with its own unique information. No two devices share the same credentials. Since a random user might want their number sent to multiple phones, and maybe even a cell phone, I've done the following: if(method=="INVITE" && !isflagset(5)) { if(avp_db_load("$ru/username","fork")) { t_on_branch("1"); avp_pushto("$ru/username","$avp(s:fork)/g"); }; }; Then the branch route looks like: branch_route[1] { setflag(5); route(3); } The idea here is the fork avp contains one or more possible destinations, but we don't store a complete URI here. The branch route sends us back through the normal routing logic to lookup local users on the system, and maybe route something to an external PSTN gateway, or another proxy. The problem: I haven't found a way to gracefully handle the event when one of the possible destinations is a device that hasn't registered. Say we allow the user to have a soft client, so we add the username to the list of fork avps. If the device isn't registered, I have 2 options: 1) Use a default route 2) Detect that I should know who this is, and error out Either way, an error is produced, but I don't see a good way of detecting if we should push the 404, or generate a 404 back to the client. The polite method would be: Are there other branches still going? If yes, just exit this branch If no, relay, or generate an error back to the client So my question is how to do this last part. Am I crazy for even thinking about doing this in OpenSER? BTW, I'm using v1.1.1 --Chris

2 1

[Users] uri module question
by Klaus Darilion 23 Feb '07

23 Feb '07

Hi! I have problems understanding the uri module. As far as I understand I can map aliases to subscribers - e.g. my main identity is klaus@mydomain, but I also want to use nickname@cooldomain in From: header with a single SIP account (single subscriber) Thus I can use uri_db module and uri table to check if klaus@mydomain is allowed to use nickname@cooldomain. Thus, what do I enter in the uri table? The definition is: id | username | domain | uri_user | last_modified ----+----------+--------+----------+--------------- In which column do I have to put which values? thanks klaus -- Klaus Darilion nic.at

2 1

[Serusers] Call from Asterisk to SER to a foreign subscriber
by karim basraoui 23 Feb '07

23 Feb '07

Hi, I've set up a small architecture in which an ASterisk proxy is connected to a SER proxy in two different domains. I have sip phones connected to both of them. when an asterisk subscriber "travels" to SER domain it tries to register to SER. I've configured SER to accept foreign subscribers and to REGISTER them to their initial domain by the command "t_replicate". Till now everything works good, the UA is registered and Asterisk is aware of its new location, it displays in peers list the UA and its new IP address and port. The problem is when i'm trying to call the foreign UA from an Asterisk subscriber, it doesn't work. i have a TRYING response but no call establishement. how can i overcome this problem please? -- BASRAOUI Karim www.basraouik.afrikart.net

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