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37468 discussions

[Kamailio-Users] HOWTO rewrite to: header
by alexander merkulov 12 Aug '09

12 Aug '09

here is correct reqwest and reqwest via kamailio is there any way to rewrite To: field in kamailio because now it nto work correct reqwest(asterisk) RECEIVING FROM: 69.70.173.195:5060 CANCEL sip:34249@193.110.78.12:8484 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.180:5060;branch=z9hG4bK4c97bd4a From: "Unknown" <sip:Unknown@voip1.bravotelecom.com<sip%3AUnknown(a)voip1.bravotelecom.com> >;tag=as366a0ba0 To: <sip:34249@193.110.78.12:8484> Call-ID: 322db75c71be4ba45a61c7666b93e6be(a)voip1.bravotelecom.com CSeq: 102 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 incorrect reqwest(multi-homed kamailio, asterisk originate to internal eth) RECEIVING FROM: 69.70.173.195:5060 CANCEL sip:34249@193.110.78.12:8484 SIP/2.0 Via: SIP/2.0/UDP 69.70.173.195;branch=z9hG4bK1b1b.3c42cea6.0 From: "Unknown" <sip:Unknown@69.70.173.195 <sip%3AUnknown(a)69.70.173.195> >;tag=as6075df60 Call-ID: 4d784fa972c4e9c9795032417f46cb26(a)voip1.bravotelecom.com To: <sip:34249@192.168.2.170 <sip%3A34249(a)192.168.2.170>> CSeq: 102 CANCEL Max-Forwards: 70 asterisk Content-Length: 0 -- Merkulov Alexander

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[Kamailio-Users] kamailio auth with service provider
by Lance Evans 12 Aug '09

12 Aug '09

Hey Guys, Apparently Kamailio cannot authenticate with my service provider. Can this not be done with a sip trunk or peering!? if so how do i setup a trunk with kamailio!? and how do i setup peering?! Thanks

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[Kamailio-Users] Fwd: Getting Error 477 while registering Softphones
by Daniel-Constantin Mierla 12 Aug '09

12 Aug '09

... fwd to list ... reply all button mismatch ... ---------- Forwarded message ---------- From: Daniel-Constantin Mierla <miconda(a)gmail.com> Date: Wed, Aug 12, 2009 at 8:27 AM Subject: Re: [Kamailio-Users] Getting Error 477 while registering Softphones To: Meftah Tayeb <tayeb.meftah(a)gmail.com> Hello, add: alias=tayeb.blogdns.com in your config file or use the domain module: http://kamailio.org/docs/modules/1.5.x/domain.html http://kamailio.org/dokuwiki/doku.php/core-cookbook:1.5.x#alias Cheers, Daniel On Wed, Aug 12, 2009 at 1:15 AM, Meftah Tayeb <tayeb.meftah(a)gmail.com>wrote: > hello, > i'm getting Error 477 while registering a Softphone from outside my domain > i added my domain, tayeb.blogdns.com to the domain name variable, > sip_domain in kamctlrc.conf > created the DB, started, and added user 1000 and 1001 > in registration: > for internal use, i use 192.168.1.3 (Kamailio IP Address) > and for external use i use my domain > for internal, is working corectly but while i register a external softphone > is getting Error 477 > what is this error message? > how i can fix it? > please try to help me, any help is welcome > > > __________ Information from ESET NOD32 Antivirus, version of virus > signature database 4327 (20090811) __________ > > The message was checked by ESET NOD32 Antivirus. > > http://www.eset.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 -- Daniel-Constantin Mierla http://www.asipto.com

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[Kamailio-Users] Getting Error 477 while registering Softphones
by Meftah Tayeb 11 Aug '09

11 Aug '09

hello, i'm getting Error 477 while registering a Softphone from outside my domain i added my domain, tayeb.blogdns.com to the domain name variable, sip_domain in kamctlrc.conf created the DB, started, and added user 1000 and 1001 in registration: for internal use, i use 192.168.1.3 (Kamailio IP Address) and for external use i use my domain for internal, is working corectly but while i register a external softphone is getting Error 477 what is this error message? how i can fix it? please try to help me, any help is welcome __________ Information from ESET NOD32 Antivirus, version of virus signature database 4327 (20090811) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com

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[Kamailio-Users] dialog info + Grandstreams = freezing
by David 11 Aug '09

11 Aug '09

Hello, I am having this problem on kamailio 1.5.2-tls compiled on Ubuntu 8.04, 1.5.1-tls ( compiled with no tls ) on Ubuntu 8.04 and with OpenSIPS 1.5.2-tls compilde on Ubuntu 8.04 I am trying to setup presence_dialoginfo with my Grandstreams, Snom and Linksys. I have a 4 phones on the server. 101 - Linksys SPA962 ( 6.1.5a ) 102 - Grandstream GXP2020 ( 1.2.1.4 ) 103 - Grandstream GXP2000 ( 1.2.1.4 ) 104 - Grandstream GXP2000 ( 1.1.6.46 ) 105 - Snom 360 ( 7.3.23 ) My Kamailio deals with registrations, NAT and BLF everything else is sent to one of two asterisk boxes. I use the dispatcher module for this. This means that when I call one extension to the other, both call legs from asterisk are going through Kamailio as separate calls. But to divide my customers, the usernames are different from the URL that the user types. For example the customer dials '101' but it is changed to testspace.101 when it comes back from asterisk. So Kamailio would have two calls in the event that 101 dials 102. sip:testspace.101@myserver to 102 ( this is sent to asterisk ) sip:testspace.101@myserver to testspace.102 ( this is coming back from asterisk ) Something is horribly wrong. I have the following problems : 1. If 102 calls 103, when 103 answers both phones hang for about 2 minutes 2. If 105 calls 101, 101 BLF comes back to the inactive state ( green on the Linksys and dark on the Snom), but the orange light stays on on the Snom and it thinks the call is still active ( the light is on, but the call is over ) 3. If any extension calls any extension and I try a call pickup, it fails. It looks like the Linksys is sending a NOTIFY to pickup the call ( I thought it was supposed to send an invite... ? ) Looking at the logs it looks like Kamailio is sending out so many NOTIFYs that it is crashing the Grandstreams, and causing the Snom to act funny. Here are some experts from my config file : root@kamailio-dev:/etc/kamailio# grep dialog * kamailio.cfg:# * avp value for dialogs is still not correct kamailio.cfg:loadmodule "dialog.so" kamailio.cfg:loadmodule "presence_dialoginfo.so" kamailio.cfg:loadmodule "pua_dialoginfo.so" kamailio.cfg:#modparam("pua_dialoginfo", "include_localremote", 0) kamailio.cfg:#modparam("pua_dialoginfo", "include_tags", 0) kamailio.cfg:#modparam("pua_dialoginfo", "include_callid", 0) kamailio.cfg:modparam("dialog", "dlg_flag", 4) kamailio.cfg:modparam("dialog", "db_mode", 1) kamailio.cfg:modparam("dialog", "timeout_avp", "$avp(i:10)") # I still haven't figured out how to set $avp(i:10) kamailio.cfg:modparam("pua_dialoginfo", "override_lifetime", 300) kamailio.cfg:modparam("presence_dialoginfo", "force_single_dialog", 1) kamailio.cfg:modparam("pua_dialoginfo", "caller_confirmed", 1) kamailio.cfg:modparam("auth_db|usrloc|acc|domain|avpops|presence|presence_xml|pua|dialog", "db_url", kamailio.cfg:# Flag 4 = Mark the current request for a dialog kamailio.cfg: # sequential request withing a dialog should the set flag looks like this : if ( ds_is_from_list() ) { xlog("L_INFO", "Coming from asterisk"); if ( is_method("INVITE")) { setflag(4); } } So the dialog flag is only set for the leg coming back from asterisk. When a notify comes in : if(is_method("NOTIFY") ) { if (! t_newtran()) { sl_reply_error(); exit; }; t_reply("200", "OK"); t_release(); exit ; } Publish and subscribe are like this : if( is_method("PUBLISH") || is_method("SUBSCRIBE") ) { route(5); exit; } route[5] { # absorb retransmissions if (! t_newtran()) { xlog("L_INFO", "Ignoring PUBLISH/SUBSCRIBE on retransmition - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_reply_error(); exit; }; append_to_reply("Contact: <sip:myserver.tld:5060>\r\n"); if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } else { } exit; } I also have NAT checking for those telephones where stun isn't enough. Before I reach publish/subscribe/invite/notify, I also call setbflag() and sometimes call fix_nated_contact(). Additionnally, I have a block if code before my presence stuff if ( has_totag() && loose_route()) { t_relay(); }. If sip.conf:canreinvite=yes, the grandstreams freeze so long that the server times out, and the BLFs get really messed up. if sip:canreinvite=no the grandstreams only freeze for about 30 seconds. Obviously I am doing something wrong, but despite having searched google for endless hours, and poured over documentation, I can not seem to find what I did wrong. I would really appreciate if someone could shed light on my problem. I am having this problem on kamailio 1.5.2-tls compiled on Ubuntu 8.04, 1.5.1-tls ( compiled with no tls ) on Ubuntu 8.04 and with OpenSIPS 1.5.2-tls compilde on Ubuntu 8.04 Thanks, David

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[Kamailio-Users] problem regarding openser+sip+raduis project
by Sanyog Kale 10 Aug '09

10 Aug '09

sir i am working on project Build SIP-based VOIP Service With RADIUS AAA Using Kamailio (OpenSER) And FreeRadius as per tutorial given by you. but the problem is 1.i am not understanding how to intialize modules for freeradius 2.while testing radius server where to make file named digest 3.the openser configuration which is given by you should be stored in which file? 4.and after all configurations done how to test all the parameters configured hoping for reply regards sanyog

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[Kamailio-Users] Traffic parsing
by michel freiha 09 Aug '09

09 Aug '09

Hi all, I would like to ask you please how to configure kamailio in order to send signaling traffic to a specific gateway and rtp traffic to another gateway? Kindly note that I'm using media proxy for rtp and carrierroute for sending traffic to different gateways Regards

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[Kamailio-Users] Siremis accounting
by Lance Evans 07 Aug '09

07 Aug '09

Hey guys Nothing seems to be showing in the Accounting tabs (in any of the subtabs) Any ideas!?

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[Kamailio-Users] Siremis not displaying anything in Online SIP Users
by Lance Evans 07 Aug '09

07 Aug '09

Hey Guys, Hope someone has time for a real newbie to VOIP, kamailio and siremis!? I got Kamailio and Siremis work ok. Fighting with Siremis accounting still... but for i cant seem see anything in the Online SIP Users when i'm making calls etc. ubuntu server 8.10 mysql v5.0.67 php v5.2.6-2 Kamailio 1.5.2 notls (built from source) + siremis 0.9.3 let me know if you guys need anything else!? Thanks

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sip-router + tls
by MÉSZÁROS Mihály 07 Aug '09

07 Aug '09

Hi All! I have on question regarding the TLS module. loadpath "/usr/lib/ser/modules" loadmodule "avpops.so" loadmodule "tls" _but_ i can use only tlsops setting parameters. modparam("tlsops","config","/etc/ser/tls.cfg") tls module name in tlsops? modules/tls_mod.c /* * Module interface */ struct module_exports exports = { "tlsops", Is this a typo or some work in progress? Is it a bug, or a feature? Thanks, Misi

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