sr-users February 2009

sr-users@lists.kamailio.org

76 participants
97 discussions

Re: [Kamailio-Users] [OpenSIPS-Users] [OpenSER-Users] [OT] How to handle different DID's in incoming calls for a registered client?

by Antonio Reale

Sorry for the delay. I did not read your reply. Nice solution. I need to check that IOS based pbx can read a custom header. Regards. Antonio. > -----Original Message----- > From: users-bounces(a)lists.opensips.org [mailto:users- > bounces(a)lists.opensips.org] On Behalf Of Iñaki Baz Castillo > Sent: Saturday, February 21, 2009 3:42 AM > To: users(a)lists.opensips.org > Subject: Re: [OpenSIPS-Users] [OpenSER-Users] [OT] How to handle different > DID's in incoming calls for a registered client? > > El Viernes, 20 de Febrero de 2009, Antonio Reale escribió: > > Hi Iñaki, > > did you find a way to solve this problem. I have the same problem with > > devices that can't read the To: value. > > For customers with 10/100 assigned number I routed the entire root number > > to the customer's IP (loosing auth...), but for users with only one > > supplementary number it is not practicable. > > Has someone a suggestion for this problem? > > Thanks to all. > > Regards. > > In the proxy I add a custom header: > P-Called-Number: +34123123123 > so the pbx receiving this INVITE can read that header. > > -- > Iñaki Baz Castillo > > _______________________________________________ > Users mailing list > Users(a)lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users

15 years, 1 month

[OpenSER-Users] [OT] How to handle different DID's in incoming calls for a registered client?

by Iñaki Baz Castillo

Hi, maybe this question is a bit off-topic so I'm sorry for that. My question is about SIP providers using OpenSer that associate PSTN numbers to their local clients (SIP accounts): Usually the client must register to OpenSer in order to receive calls. Then it will appear in location table with "Contact=sip:clientX@IP". Suppose clientX has two PSTN numbers associated in a ENUM entry: +34999000111 +34999000222 When anyone in PSTN world calls to +34999000222 the call will arrive to the OpenSer from a gateway in an INVITE like: INVITE sip:+34999000222@gateway SIP/2.0 To: <sip:+34999000222@gateway> The OpenSer will do the lookup in location table and finally send this INVITE to the clientX: INVITE sip:clientX@IP_clientX SIP/2.0 To: <sip:+34999000222@gateway> The info about the called PSTN number is just available in "To" header, so a way to get different behaviour for each associated PSTN number is matching "To" URI. Is common to do it? which other alternatives are there? Thanks for any comment. Regards. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

15 years, 1 month

Re: [Kamailio-Users] ACC with mysql

by Daniel-Constantin Mierla

According to the config, you do not even set the accounting flag for ACK, or I missed something there? Can you compile with memory debug turned on and test? Watch the syslog for critical errors. It can turn to be a buffer overflow somewhere. Once again, what is the version are you using? Cheers, Daniel On 02/27/2009 01:45 PM, Uriel Rozenbaum wrote: > Here it is > > On Fri, Feb 27, 2009 at 9:31 AM, Daniel-Constantin Mierla > <miconda(a)gmail.com <mailto:miconda@gmail.com>> wrote: > > > > On 02/27/2009 12:57 PM, Uriel Rozenbaum wrote: > > I´ve checked the captures again and all ACK messages have the > RPID, for 487 or 200 messages. > > ok, but is the ack you sent causing the trouble? It does not look > as an ack for 200ok, because it does not have a Route header. > Non-200ok ACK do not get accounted. > > > I'm just using a plain build of kamailio downloaded from the > site including carrerroute module. > > Would it be useful if I sent the .cfg? > > > Could help, yes. > > Daniel > > > On Fri, Feb 27, 2009 at 8:34 AM, Daniel-Constantin Mierla > <miconda(a)gmail.com <mailto:miconda@gmail.com> > <mailto:miconda@gmail.com <mailto:miconda@gmail.com>>> wrote: > > I tried to reproduce, but no success. The crash is > indicated when > processing the lumps, a very old piece of code, so I guess > there > is somewhere a buffer overlow. > > Are you having some private modules that are you using the > config > file? > > Thanks, > Daniel > > > > On 02/27/2009 11:39 AM, Daniel-Constantin Mierla wrote: > > Hello, > > is this of an ACK due to 200ok? Are you doing > record-routing? > Because Route header is missing. > > If you don't do record-routing, then the ACK should be > end-to-end between caller and callee and you cannot > account. > > Anyway, the reported issue should be fixed. > > Cheers, > Daniel > > > On 02/24/2009 10:05 PM, Uriel Rozenbaum wrote: > > Daniel, I've been trying a litte longer with the server > and I captured one ACK that definetely has RPID: > > /ACK sip:1054111556446967@cc1int.x.com.ar > <mailto:sip%3A1054111556446967@cc1int.x.com.ar> > <mailto:sip%3A1054111556446967@cc1int.x.com.ar > <mailto:sip%253A1054111556446967@cc1int.x.com.ar>> > <mailto:sip%3A1054111556446967@cc1int.x.com.ar > <mailto:sip%253A1054111556446967@cc1int.x.com.ar> > <mailto:sip%253A1054111556446967@cc1int.x.com.ar > <mailto:sip%25253A1054111556446967@cc1int.x.com.ar>>> SIP/2.0 > > Via: SIP/2.0/UDP > 192.168.200.11:5060;branch=z9hG4bK31a69f48;rport > From: "541160911100" > <sip:541160911100@192.168.200.11 > <mailto:sip%3A541160911100@192.168.200.11> > <mailto:sip%3A541160911100@192.168.200.11 > <mailto:sip%253A541160911100@192.168.200.11>> > <mailto:sip%3A541160911100@192.168.200.11 > <mailto:sip%253A541160911100@192.168.200.11> > <mailto:sip%253A541160911100@192.168.200.11 > <mailto:sip%25253A541160911100@192.168.200.11>>>>;tag=as799b3334 > > To: <sip:1054111556446967@cc1int.cpsnet.com.ar > <mailto:sip%3A1054111556446967@cc1int.cpsnet.com.ar> > <mailto:sip%3A1054111556446967@cc1int.cpsnet.com.ar > <mailto:sip%253A1054111556446967@cc1int.cpsnet.com.ar>> > <mailto:sip%3A1054111556446967@cc1int.cpsnet.com.ar > <mailto:sip%253A1054111556446967@cc1int.cpsnet.com.ar> > > <mailto:sip%253A1054111556446967@cc1int.cpsnet.com.ar > <mailto:sip%25253A1054111556446967@cc1int.cpsnet.com.ar>>>>;tag=f8f2ab2c1295e90ed7dbb499b30f44b2.90f4 > > > > Contact: <sip:541160911100@192.168.200.11 > <mailto:sip%3A541160911100@192.168.200.11> > <mailto:sip%3A541160911100@192.168.200.11 > <mailto:sip%253A541160911100@192.168.200.11>> > <mailto:sip%3A541160911100@192.168.200.11 > <mailto:sip%253A541160911100@192.168.200.11> > <mailto:sip%253A541160911100@192.168.200.11 > <mailto:sip%25253A541160911100@192.168.200.11>>>> > > Call-ID: > 5d7adcce1399bafd45c194b049479a1b(a)192.168.200.11 > <mailto:5d7adcce1399bafd45c194b049479a1b@192.168.200.11> > > <mailto:5d7adcce1399bafd45c194b049479a1b@192.168.200.11 > <mailto:5d7adcce1399bafd45c194b049479a1b@192.168.200.11>> > > <mailto:5d7adcce1399bafd45c194b049479a1b@192.168.200.11 > <mailto:5d7adcce1399bafd45c194b049479a1b@192.168.200.11> > > <mailto:5d7adcce1399bafd45c194b049479a1b@192.168.200.11 > <mailto:5d7adcce1399bafd45c194b049479a1b@192.168.200.11>>> > CSeq: 102 ACK > User-Agent: Metrotel > Max-Forwards: 70 > Remote-Party-ID: "541160911100" > <sip:541160911100@192.168.200.11 > <mailto:sip%3A541160911100@192.168.200.11> > <mailto:sip%3A541160911100@192.168.200.11 > <mailto:sip%253A541160911100@192.168.200.11>> > <mailto:sip%3A541160911100@192.168.200.11 > <mailto:sip%253A541160911100@192.168.200.11> > <mailto:sip%253A541160911100@192.168.200.11 > <mailto:sip%25253A541160911100@192.168.200.11>>>>;privacy=off;screen=no > > > Content-Length: 0/ > > Hope it helps, > Uriel > > > -- > Daniel-Constantin Mierla > http://www.asipto.com > > -- Daniel-Constantin Mierla http://www.asipto.com

15 years, 1 month

[Kamailio-Users] ACC with mysql

by Uriel Rozenbaum

Hey guys, I'm using kamailio-1.4.3-notls with ACC module on mysql. I'm trying to benchmark it installed on a virtual server (Xen on CentOS) and I have everything going OK for my basic configuration but when I turn on ACC (by flagging transactions) I get the following lines on the log: Feb 19 08:48:57 ops3 /usr/local/sbin/kamailio[4895]: INFO:core:handle_sigs: child process 4899 exited by a signal 11 Feb 19 08:48:58 ops3 /usr/local/sbin/kamailio[4895]: INFO:core:handle_sigs: core was generated Feb 19 08:48:58 ops3 /usr/local/sbin/kamailio[4895]: INFO:core:handle_sigs: terminating due to SIGCHLD Feb 19 08:48:58 ops3 /usr/local/sbin/kamailio[4902]: INFO:core:sig_usr: signal 15 received Feb 19 08:48:58 ops3 /usr/local/sbin/kamailio[4903]: INFO:core:sig_usr: signal 15 received Feb 19 08:48:58 ops3 /usr/local/sbin/kamailio[4901]: INFO:core:sig_usr: signal 15 received Feb 19 08:48:58 ops3 /usr/local/sbin/kamailio[4897]: INFO:core:sig_usr: signal 15 received Feb 19 08:48:58 ops3 /usr/local/sbin/kamailio[4904]: INFO:core:sig_usr: signal 15 received Feb 19 08:48:58 ops3 /usr/local/sbin/kamailio[4898]: INFO:core:sig_usr: signal 15 received This issue only happens when i account messages using the following lines: if (is_method("INVITE|BYE|CANCEL")) { setflag(1); setflag(2); } and the configuration for the modules are ass follows: # ----- acc params ----- /* what sepcial events should be accounted ? */ modparam("acc", "early_media", 1) modparam("acc", "report_ack", 1) modparam("acc", "report_cancels", 1) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "failed_transaction_flag", 3) modparam("acc", "log_flag", 1) modparam("acc", "log_missed_flag", 2) /* uncomment the following lines to enable DB accounting also */ modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 2) # Descomentar para guardar parametros adicionales # modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;dst_user=$rU;dst_domain=$rd") modparam("acc", "db_table_acc", "acc") modparam("acc", "db_table_missed_calls", "missed_calls") modparam("acc", "db_url", "mysql://kamailio:kamailiorw@localhost/kamailio") modparam("acc", "acc_method_column", "method") modparam("acc", "acc_from_tag_column", "from_tag") modparam("acc", "acc_to_tag_column", "to_tag") modparam("acc", "acc_callid_column", "callid") modparam("acc", "acc_sip_code_column", "sip_code") modparam("acc", "acc_sip_reason_column", "sip_reason") modparam("acc", "acc_time_column", "time") modparam("acc", "db_extra", "cseq=$cs;privacy=$dip;reason=$dir;from_uri=$fu;from_username=$fU;from_displayname=$fn;from_domain=$fd;req_username=$rU;req_domain=$rd;rpid=$re;source_ip_addr=$si;source_port=$sp;to_uri=$tu;to_username=$tU;to_displayname=$tn;to_domain=$td;original_uri=$ou;original_uri_username=$oU;original_uri_domain=$od;unix_timestamp=$Ts") Do you think there is something I'm missing on the configuration? I tried to modify parameters on mysql but get the same errors all the time? Thanks in advance! Uriel

15 years, 1 month

Re: [Kamailio-Users] how to know if the next gw is up?

by Daniel-Constantin Mierla

Please direct your questions to users mailing lists. Private messages of such type are not honored very fast, they usually end in my very long unread mails pool. See the dispatcher module, maybe it helps you. Thanks, Daniel On 02/27/2009 11:18 AM, BERGANZ François wrote: > Hello, > > I have 1SER and 5 Asterisk. > I need to choose the dst with my algo... > But, I need to have my server list and choose a server which is online. > The best thing could be to have OPTIONS send to my asterisk and write responses in the database! > > Have you an idea? > > > Cordialement, > BERGANZ François >  Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. > > -----Message d'origine----- > De : users-bounces(a)lists.kamailio.org [mailto:users-bounces@lists.kamailio.org] De la part de Daniel-Constantin Mierla > Envoyé : vendredi 27 février 2009 09:10 > À : c_lougher(a)yahoo.co.uk > Cc : users(a)lists.kamailio.org > Objet : Re: [Kamailio-Users] Kamailio Newb questions > > > > On 02/27/2009 12:07 AM, carl Lougher wrote: > >> So if i just want to hook off the rtp stream between endpoint and sip provider then am i better off to go for a stun server than nathelper/rtpproxy? >> >> > nathelper/rtpproxy does rtp relay on server, so the rtp goes: [caller] > === [rtpproxy] === [callee] > > In case of dealing with symmetric NAT, it is the only feasible way now > to make it work. > > If STUN is used, the rtp goes: [caller] === [callee] . But does not work > for symmetric nat. > > The best is to combine both of them so you get as less as possible RTP > relaying on server. > > Cheers, > Daniel > > >> What are you using rtpproxy for that is different than stun? >> >> >> --- On Thu, 26/2/09, Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: >> >> >> >>> From: Daniel-Constantin Mierla <miconda(a)gmail.com> >>> Subject: Re: [Kamailio-Users] Kamailio Newb questions >>> To: c_lougher(a)yahoo.co.uk >>> Cc: users(a)lists.kamailio.org >>> Date: Thursday, 26 February, 2009, 12:27 PM >>> On 02/26/2009 01:19 PM, carl Lougher wrote: >>> >>> >>>> Thanks for that. So does it mean by using rtpproxy you >>>> >>>> >>> will therefore carry all the rtp streams through that server >>> yes, that is the role of RTPProxy - to proxy the RTP >>> streams, therefore those go via the server. >>> >>> If you want end-to-end RTP stream, then look at STUN, if >>> the phones are not behind symmetric nat, it can help. >>> >>> >>> >>>> or can it be redirected to the sip provider from the >>>> >>>> >>> endpoint? >>> >>> >>>> Also how do you put the kamailio server in the >>>> >>>> >>> equation? Do you set it up as an external proxy for the >>> clients or do you register the clients to it then just use >>> asterisk for the media/vmail etc? >>> >>> >>>> >>>> >>>> >>> I do everything in kamailio but the media services which i >>> do with asterisk (vmail, ivr, ...) - authentication, >>> registration, call routing is done in kamailio. >>> >>> Cheers, >>> Daniel >>> >>> >>> >>>> --- On Thu, 26/2/09, Daniel-Constantin Mierla >>>> >>>> >>> <miconda(a)gmail.com> wrote: >>> >>> >>>> >>>> >>>> >>>>> From: Daniel-Constantin Mierla >>>>> >>>>> >>> <miconda(a)gmail.com> >>> >>> >>>>> Subject: Re: [Kamailio-Users] Kamailio Newb >>>>> >>>>> >>> questions >>> >>> >>>>> To: c_lougher(a)yahoo.co.uk >>>>> Cc: users(a)lists.kamailio.org >>>>> Date: Thursday, 26 February, 2009, 9:16 AM >>>>> Hello, >>>>> >>>>> On 02/26/2009 12:59 AM, carl Lougher wrote: >>>>> >>>>> >>>>> >>>>>> Howdy, >>>>>> I'm trying to remove the media/rtp streams >>>>>> >>>>>> >>> from an >>> >>> >>>>>> >>>>>> >>>>>> >>>>> asterisk server for natted users so would like to >>>>> >>>>> >>> know if >>> >>> >>>>> this is possible with kamailio. >>>>> >>>>> >>>>> >>>>>> >>>>>> >>>>>> >>>>> yes it is possible. nathelper+rtpproxy is the >>>>> >>>>> >>> option I use >>> >>> >>>>> and prefer because of flexibility and >>>>> >>>>> >>> performances. You can see an >>> >>> >>>>> example at: >>>>> >>>>> >>>>> >>> http://www.voip-info.org/wiki/view/Kamailio+1.5.x+and+RTPProxy >>> >>> >>>>> >>>>> >>>>> >>>>>> Qu's: >>>>>> What is the best option? >>>>>> rtpproxy/mediaproxy? >>>>>> nathelper? >>>>>> >>>>>> If i use kamailio to achieve this does it mean >>>>>> >>>>>> >>> that i >>> >>> >>>>>> >>>>>> >>>>>> >>>>> still have to carry the rtp streams through the >>>>> >>>>> >>> kamailio >>> >>> >>>>> server instead? >>>>> >>>>> >>>>> >>>>>> >>>>>> >>>>>> >>>>> through the rtpproxy server, which can be located >>>>> >>>>> >>> on same >>> >>> >>>>> or different machine than kamailio. >>>>> >>>>> >>>>> >>>>> >>>>>> Also will i need to change the logon info for >>>>>> >>>>>> >>> the >>> >>> >>>>>> >>>>>> >>>>>> >>>>> clients so they now logon to kamailio then i just >>>>> >>>>> >>> point >>> >>> >>>>> registrar to asterisk? >>>>> >>>>> >>>>> >>>>>> Can i use kamailio for sip trunks to asterisk >>>>>> >>>>>> >>> and >>> >>> >>>>>> >>>>>> >>>>>> >>>>> carry rtp and natted clients media streams rather >>>>> >>>>> >>> than >>> >>> >>>>> register to asterisk? >>>>> >>>>> >>>>> >>>>>> >>>>>> >>>>>> >>>>> Yes, you can register to kamailio, see registrar >>>>> >>>>> >>> and usrloc >>> >>> >>>>> modules. >>>>> >>>>> Cheers, >>>>> Daniel >>>>> >>>>> >>>>> >>>>> >>>>>> Many thanks, >>>>>> Taff.. >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>> _______________________________________________ >>> >>> >>>>>> Kamailio (OpenSER) - Users mailing list >>>>>> Users(a)lists.kamailio.org >>>>>> >>>>>> >>>>>> >>>>>> >>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >>> >>> >>>>> >>>>> >>>>> >>> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users >>> >>> >>>>> >>>>> >>>>> >>>>>> >>>>>> >>>>>> >>>>> -- Daniel-Constantin Mierla >>>>> http://www.asipto.com >>>>> >>>>> >>>>> >>>> >>>> _______________________________________________ >>>> Kamailio (OpenSER) - Users mailing list >>>> Users(a)lists.kamailio.org >>>> >>>> >>>> >>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >>> >>> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users >>> >>> >>>> >>>> >>>> >>> -- Daniel-Constantin Mierla >>> http://www.asipto.com >>> >>> >> >> >> _______________________________________________ >> Kamailio (OpenSER) - Users mailing list >> Users(a)lists.kamailio.org >> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users >> >> >> > > -- Daniel-Constantin Mierla http://www.asipto.com

15 years, 1 month

[Kamailio-Users] Kamailio Newb questions

by carl Lougher

Howdy, I'm trying to remove the media/rtp streams from an asterisk server for natted users so would like to know if this is possible with kamailio. Qu's: What is the best option? rtpproxy/mediaproxy? nathelper? If i use kamailio to achieve this does it mean that i still have to carry the rtp streams through the kamailio server instead? Also will i need to change the logon info for the clients so they now logon to kamailio then i just point registrar to asterisk? Can i use kamailio for sip trunks to asterisk and carry rtp and natted clients media streams rather than register to asterisk? Many thanks, Taff..

15 years, 1 month

[Kamailio-Users] default config file

by Elena-Ramona Modroiu

Hi, looking at the default config file, we deliver something pretty useless and very insecure. I propose to add to it several features so it becomes really usable: - mysql support - even some use radius for aaa, mysql is for sure used for usrloc. Those using other db drivers can just replace the module - auth against db and check auth username against from/to username to avoid spoofing - accounting - at least to syslog to store more than default data which gives nothing useful right now. Thinking to from/ruri username and domain at least in order to have caller and callee details - usrloc to database - nat traversal support with nathelper and rtpproxy - if rtpproxy is not running, will be just warnings at startup - the first part of the config will have some help of how to configure the environment to start the sip server (e.g., add user, ...) What do you think? Regards, Ramona

15 years, 1 month

[Serusers] Opensips--Ser

by Parikhita Behera

Dear Sir I am new to SER . So I want to know , is Opensips is Open source?. Does it need license?. What operating system needed to install the SER. Does SER support totally telephony, pc-pc , pc-phone, calling card , call back , IPPBX etc.. How many systems needed to install to work the complete solution. If possible , you can give me the installation and configuration ducumentation so that I can try to do the complete solution. Waiting for your reply. Thanks and Regards Parikhita Add more friends to your messenger and enjoy! Go to http://messenger.yahoo.com/invite/

15 years, 1 month

[Serusers] Need easiest way to set/clear a flag and refer to it later in same call session

by Frank Durda IV

I need a simple way to set either a boolean flag (or integer value) during the INVITE of a given call in ser.cfg, and test that value via "if" elsehere in ser.cfg during the 183/200 response messages for that same call. This would be a data object unique to a given call, which should disappear/lose persistence/end scope when a given call ends. Not global, no need to write the value to disk, doesn't need to survive a restart of SER, etc. The various SER documents describe a number of things that could do this, but most seem overly complex, storing the object in a MYSQL database on disk or something, and these all seem so excessive. There also appear to be a few built-in flags, but the documentation suggests that all are actually there for specific uses and so use for other things might not work. I simply want to make a determination of something during the INVITE for call 12345, and be able to refer to that flag or integer value later on when the various response messages for the same call (#12345) are coming back through. I can't make the same determination on each message after the INVITE because the original src_ip address value where the INVITE came from doesn't seem to be in a testable variable when handling reply messages, or at least not in one that is clearly documented. I've chased a number of possible ways this might be done, but tire of being referred away from the doc that came with the source and over to web sites that tell me this or that module that might do part of what I need is deprecated and to use this other module, but the web site for the other module says that the first module is better suited to what I want to do (a circle), doesn't show how the set value can actually be tested, or the web page just doesn't work anymore. So, assuming there is a simple way to do this, I need a quick code snip showing how to set/clear/assign a value to such a beast within INVITE handling, and a quick code snip of how to test it in an "if" statement during 183/200/110 type response messages. Whatever it is, the value set, cleared or being tested in one call should not affect any other call. Thanks in advance!

15 years, 1 month

[Serusers] Opensips--Ser

by Parikhita Behera

Dear Sir I am new to SER . So I want to know , is Opensips is Open source?. Does it need license?.. What operating system needed to install the SER. Does SER support totally telephony, pc-pc , pc-phone, calling card , call back , IPPBX etc.. How many systems needed to install to work the complete solution. If possible , you can give me the installation and configuration ducumentation so that I can try to do the complete solution. Waiting for your reply. Thanks and Regards Parikhita Add more friends to your messenger and enjoy! Go to http://messenger.yahoo.com/invite/

15 years, 1 month

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sr-users February 2009