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37470 discussions

[Kamailio-Users] Select a 3XX branch inmediatelly and CANCEL a ringing branch
by Iñaki Baz Castillo 18 Mar '09

18 Mar '09

Hi, let's me explain a simple escenario and the behaviour I want: Kamailio phone1 phone2-1 phone2-2 phone3 (201) (202) (202) (203) 1) 201 calls 202 so both phone2-1 and phone2-2 start ringing (180) in parallel. 2) phone2-1 replies a redirect "302" to user 203. 3) Kamailio cancels the branch with phone2-1 and replies 302 upstream *inmediatelly*. This is however not the most "compliant" behaviour, since Kamailio should wait for phone2-1 to reply a final response and then choose which one response to reply upstream. And that is of course the default behaviour in Kamailio. Is it possible however the behaviour I want? I find no way to get it. Thanks. -- Iñaki Baz Castillo <ibc(a)aliax.net>

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[Kamailio-Users] Hor to strip() $rU dynamically
by Iñaki Baz Castillo 18 Mar '09

18 Mar '09

Hi, I receive INVITE like this: INVITE sip:client_xxx2004@domain.org SIP/2.0 and I get from an $avp(CLIENT) the value "client_xxx". I need to strip from $rU the lenght of $avp(CLIENT), this is: # Original $rU => client_xxx2004 strip($avp(CLIENT)); # New $rU => 2004 Unfortunatelly "strip()" doesn't accept a PV as parameter, but a static integer. Any suggestion on how to achieve my goal? Thanks a lot. -- Iñaki Baz Castillo <ibc(a)aliax.net>

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[Kamailio-Users] I don't want "rpid" column but get an ERROR in "auth_db"
by Iñaki Baz Castillo 18 Mar '09

18 Mar '09

Hi, I don't want/need the "rpid" column in "subscriber" table but if I delete it I get an error: ------- Mar 18 14:49:47 [22377] ERROR:db_mysql:db_mysql_submit_query: driver error on query: Unknown column 'rpid' in 'field list' Mar 18 14:49:47 [22377] ERROR:core:db_do_query: error while submitting query Mar 18 14:49:47 [22377] ERROR:auth_db:get_ha1: failed to query database ------- After readin "auth" module documentation I've found a parameter: ------------ rpid_avp (string) Full AVP specification for the AVP which stores the RPID value. It used to transport the RPID value from authentication backend modules (auth_db or auth_radius) or from script to the auth function append_rpid_hf and is_rpid_user_e164. If defined to NULL string, all RPID functions will fail at runtime. ------------ I expected that if I set: modparam("auth", "rpid_avp", "NULL") then "auth_db" module wouldn't require "rpid" column, but I must be wrong since I get an error: ------ ERROR:auth:init_rpid_avp: malformed or non AVP NULL AVP definition ------ So, is there any way to avoid "rpid" column usage? Thanks. -- Iñaki Baz Castillo <ibc(a)aliax.net>

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Re: [Kamailio-Users] Using Asterisk as media server
by Mark Sayer 18 Mar '09

18 Mar '09

Pickup with OpenSER >> Asterisk 1.4.17 works just fine. Mark At 12:26 p.m. 18/03/2009, you wrote: >No, I don't need app_pickup... but thanks for the heads up. > >On Tue, Mar 17, 2009 at 8:21 PM, Iñaki Baz Castillo <ibc(a)aliax.net> wrote: > > El Miércoles, 18 de Marzo de 2009, Geoffrey Mina escribió: > >> I tend to not agree with some of what people are saying here. We have > >> deployed a Kamailio/Asterisk platform with GREAT success. Although, I > >> have the luxury of having many asterisk servers to achieve a common > >> goal. I have found asterisk to be VERY stable, > > > >> I am running 1.4.23. > > > > Then I hope you don't need Asterisk pickup code (since it doesn't work in > > 1.4.23 due to a regression). > > > > XD > > > > > > -- > > Iñaki Baz Castillo > > > > _______________________________________________ > > 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 > > > >_______________________________________________ >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

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[Kamailio-Users] The hash_table_2 global variable is defined twice?
by catalina oancea 18 Mar '09

18 Mar '09

Hi, I have a question regarding the global hash_table_2 variable. It seems to me that it is defined twice in different modules: ./modules/permissions/trusted.c:47:struct trusted_list **hash_table_2; /* Pointer to hash table 2 */ ./modules/domain/domain_mod.c:85:struct domain_list **hash_table_2 = 0; /* Pointer to hash table 2 */ Could you please explain why this variable is defined twice?

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[Kamailio-Users] Dialog SNMP stats.
by Alex Balashov 18 Mar '09

18 Mar '09

Is it possible to monitor via SNMPStats the amount of dialogs currently open in a particular profile? -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775

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Re: [Kamailio-Users] Using Asterisk as media server
by Juan Perez 18 Mar '09

18 Mar '09

Alex, I understand your point and appreciate your input highly, but let's say that asterisk is needed as the media server for other reasons. It may not scale as well as the other solutions but we need them to be there to provide other services. I just wanted to know whether there was any difference aside from the scalabilty issue. We were under the impression that it was imperative to use either media-proxy or rtp-proxy but after examining in more detail how the 2 solutions work we find no reason not to use asterisk as the media proxy as well. Unless of course we are missing the big picture here and surely will find out once we set out to test it. But given the deep knowledge of a lot of people in this forum we deemed necessary to ask about the feasibility of this workaround. If somebody detects any flaw on this solution or if there is someting else critical or not that we are missing we would love to hear it . thank you in advance and thank you Alex again, jp ----- Original Message ---- From: Alex Balashov <abalashov(a)evaristesys.com> To: Juan Perez <jperezsip2008(a)yahoo.com> Sent: Monday, March 16, 2009 6:51:34 PM Subject: Re: [Kamailio-Users] Using Asterisk as media server No, because the amount of throughput you get per media/RTP proxy is much higher than you would get through something whose entire core is structured to behave as a PBX, with all the attendant features and latent event loop characteristics. A media relay does one thing and one thing only, and it does it well. Asterisk does many things, among them media relay. So, while you could use Asterisk in this capacity, I don't see the point. It'd be a waste of time. Also, you don't really want to get Asterisk involved in the signaling layer (as a B2BUA) if you can help it. It introduces unnecessary complexity and possible points of interop problems and failure. With a media relay that is controlled by API hooks into Kamailio, you don't have that problem. On Mon, 16 Mar 2009 15:22:27 -0700 (PDT), Juan Perez <jperezsip2008(a)yahoo.com> wrote: > > thanks alex, but actually there will be several asterisk servers, managed > by the LCR module, that should be OK right? > it should be the same as adding several media-proxies or rtp-proxies, am I > wrong? > > > > > > > > ----- Original Message ---- > From: Alex Balashov <abalashov(a)evaristesys.com> > To: Juan Perez <jperezsip2008(a)yahoo.com> > Cc: users(a)lists.kamailio.org > Sent: Monday, March 16, 2009 3:41:46 PM > Subject: Re: [Kamailio-Users] Using Asterisk as media server > > No, nothing needs to be modified. > > However, in this scenario you are limited to Asterisk's performance > characteristics in terms of the number of concurrent calls that can be > handled. Just about any media relay will be much more scalable and > lightweight. > > Asterisk is a PBX and a nice feature server. You don't build SIP service > delivery platforms out of Asterisk used as a general-purpose call router. > > Juan Perez wrote: > >> Hello >> >> In this scenario, customers behind NAT register in kamailio, records are > stored in a db etc. Kamailio uses Asterisk boxes as gateways to send calls > to PSTN. >> Is it possible to use Asterisk as the media server instead of > media-proxy or rtp-proxy? >> In this case the SDP that Asterisk sends in the 180 message needs to me > modified? >> >> thank you >> jp >> >> >> >> >> _______________________________________________ >> 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 > > > -- Alex Balashov > Evariste Systems > Web : http://www.evaristesys.com/ > Tel : (+1) (678) 954-0670 > Direct : (+1) (678) 954-0671 > Mobile : (+1) (678) 237-1775 -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775

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[Serusers] Record-Route issues
by Stephen Helfen 17 Mar '09

17 Mar '09

I have a 0.9.6 SER proxy running and one particular UA that is creating some problems with a RECORD-ROUTE header I am not handling correctly. Here is how I have my record_route configured (XXXs are replaced by my actual proxy IP): if (!(uri=~^"sip:XXX.XXX.XXX.XXX")) { record_route_preset("XXX.XXX.XXX.XXX:5060;nat=yes"); }; When inbound traffic arrives for this particular UA, the INVITE is passed along. when the 200 OK comes back, there is a Record-Route header that looks like this (YYYs are replaced with the UA's public IP): RECORD-ROUTE: <sip:YYY.YYY.YYY.YYY:5060;lr;transport=udp> Instead of adding a new Record-Route header with my IP before this one, the message is simply proxied back to the call origination. The ACK is then sent DIRECTLY to the UA, bipassing my proxy...as well as the BYE/200 OK that terminates the call....thus rendering my stateful proxy as stateless. Can I remove the RECORD-ROUTE that my terminating UA is attaching? If not, why am I not adding my own Record-Route header like normal? What am I missing? Thanks, Steve _________________________________________________________________ Windows Live™ Groups: Create an online spot for your favorite groups to meet. http://windowslive.com/online/groups?ocid=TXT_TAGLM_WL_groups_032009

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[Kamailio-Users] 487 Request Terminated and BYE
by Guillaume Lacroix 17 Mar '09

17 Mar '09

Hi, I am facing problem with handling CANCEL/487 Request Terminated in a case of a call hunting. The problem is that the call terminating carrier doesn't process the ACK properly and keeps sending 487 messages. Here is an example. Let's say UAC1 has the following rules of call hunting : 1. call +33123456 for 10s 2. call +33789456 for 10s Let's say that an incoming call to UAC1 triggers the call hunting. Here is a diagram : openSER -> Terminating Carrier (TC) : INVITE +33123456 ... (time out : +33123456 didn't answer in 10s, so openSER sends an INVITE to +33789456 - and let's say to an other Terminating carrier TC2 - and CANCEL the current INVITE to TC) openSER -> TC2 : INVITE : +33789456 openSER -> TC : CANCEL TC -> openSER : 487 Request Terminated openSER -> TC : ACK The problem is now TC doesn't process the ACK correctly and keeps sending 487. So, in the case of +33789456 answering the call (a 200 OK is sent to openSER), openSER will keep relaying the 487 to TC2 and TC2 will then send a BYE a terminate the call : TC -> openSER : 487 openSER -> TC : ACK TC -> openSER : 487 openSER -> TC : ACK TC2 -> openSER : OK openSER -> TC2 : ACK <-- The call is taking place --> TC -> openSER : 487 openSER -> TC : ACK openser -> TC2 : 487 (openSER relays the 487 once the call has been established to TC2) TC2 -> openSER : ACK TC2 -> openSER : BYE <-- Call is ended but should not --> According to the RFC, once a call has been OKed and a 487 is received, TC2 may go on with the call or send a BYE (up to it). So it behaves the right way (chapter 15. end of 3rd paragraph : "If the INVITE results in 2xx final response(s) to the INVITE, this means that a UAS accepted the invitation while the CANCEL was in progress. The UAC MAY continue with the sessions established by any 2xx responses, or MAY terminate them with BYE."). My question is then : is there a way to prevent this behavior when a terminating carrier doesn't behave correctly, either by preventing relaying of the 487 once it has been ACKed or once the call has been OKed (but I guess we are not RFC compliant then) ? Thanks Bye, Guillaume

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[Kamailio-Users] kamailio at FFG 2009
by Henning Westerholt 17 Mar '09

17 Mar '09

Hi all, i'll present in the next week the Kamailio project at the FFG 2009 conference. Topics of my talk are: - overview about Kamailio - applications and usage scenarios - new and interesting stuff in the 1.5.0 release - future developments, the sip-router project Date: Thursday, 12.03.2009 Place: Karlsruhe, Germany Link to the abstract (in german as the talk will be also held in german): http://www.guug.de/veranstaltungen/ffg2009/abstracts.html#3_2_1 More information about the event: http://www.guug.de/veranstaltungen/ffg2009/ I'll provide the slides and eventual some notes after the talk as usual. Cheers, Henning

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